webrtc_m130/pc/jsep_transport_controller_unittest.cc

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/*
* Copyright 2018 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/jsep_transport_controller.h"
#include <map>
#include <memory>
#include "api/test/fake_media_transport.h"
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
#include "api/test/loopback_media_transport.h"
#include "api/transport/media/media_transport_interface.h"
#include "p2p/base/fake_dtls_transport.h"
#include "p2p/base/fake_ice_transport.h"
#include "p2p/base/no_op_dtls_transport.h"
#include "p2p/base/transport_factory_interface.h"
#include "p2p/base/transport_info.h"
#include "rtc_base/gunit.h"
#include "rtc_base/thread.h"
#include "test/gtest.h"
using cricket::Candidate;
using cricket::Candidates;
using cricket::FakeDtlsTransport;
using webrtc::SdpType;
static const int kTimeout = 100;
static const char kIceUfrag1[] = "u0001";
static const char kIcePwd1[] = "TESTICEPWD00000000000001";
static const char kIceUfrag2[] = "u0002";
static const char kIcePwd2[] = "TESTICEPWD00000000000002";
static const char kIceUfrag3[] = "u0003";
static const char kIcePwd3[] = "TESTICEPWD00000000000003";
static const char kAudioMid1[] = "audio1";
static const char kAudioMid2[] = "audio2";
static const char kVideoMid1[] = "video1";
static const char kVideoMid2[] = "video2";
static const char kDataMid1[] = "data1";
namespace webrtc {
namespace {
// Media transport factory requires crypto settings to be present in order to
// create media transport.
void AddCryptoSettings(cricket::SessionDescription* description) {
for (auto& content : description->contents()) {
content.media_description()->AddCrypto(cricket::CryptoParams(
/*t=*/0, std::string(rtc::CS_AES_CM_128_HMAC_SHA1_80),
"inline:YUJDZGVmZ2hpSktMbW9QUXJzVHVWd3l6MTIzNDU2", ""));
}
}
} // namespace
class FakeTransportFactory : public cricket::TransportFactoryInterface {
public:
std::unique_ptr<cricket::IceTransportInternal> CreateIceTransport(
const std::string& transport_name,
int component) override {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
return std::make_unique<cricket::FakeIceTransport>(transport_name,
component);
}
std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
cricket::IceTransportInternal* ice,
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
const webrtc::CryptoOptions& crypto_options) override {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
return std::make_unique<FakeDtlsTransport>(
static_cast<cricket::FakeIceTransport*>(ice));
}
};
class JsepTransportControllerTest : public JsepTransportController::Observer,
public ::testing::Test,
public sigslot::has_slots<> {
public:
JsepTransportControllerTest() : signaling_thread_(rtc::Thread::Current()) {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
fake_transport_factory_ = std::make_unique<FakeTransportFactory>();
}
void CreateJsepTransportController(
JsepTransportController::Config config,
rtc::Thread* signaling_thread = rtc::Thread::Current(),
rtc::Thread* network_thread = rtc::Thread::Current(),
cricket::PortAllocator* port_allocator = nullptr) {
config.transport_observer = this;
config.rtcp_handler = [](const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us) { RTC_NOTREACHED(); };
// The tests only works with |fake_transport_factory|;
config.external_transport_factory = fake_transport_factory_.get();
// TODO(zstein): Provide an AsyncResolverFactory once it is required.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
transport_controller_ = std::make_unique<JsepTransportController>(
signaling_thread, network_thread, port_allocator, nullptr, config);
ConnectTransportControllerSignals();
}
void ConnectTransportControllerSignals() {
transport_controller_->SignalIceConnectionState.connect(
this, &JsepTransportControllerTest::OnConnectionState);
transport_controller_->SignalStandardizedIceConnectionState.connect(
this, &JsepTransportControllerTest::OnStandardizedIceConnectionState);
transport_controller_->SignalConnectionState.connect(
this, &JsepTransportControllerTest::OnCombinedConnectionState);
transport_controller_->SignalIceGatheringState.connect(
this, &JsepTransportControllerTest::OnGatheringState);
transport_controller_->SignalIceCandidatesGathered.connect(
this, &JsepTransportControllerTest::OnCandidatesGathered);
}
std::unique_ptr<cricket::SessionDescription>
CreateSessionDescriptionWithoutBundle() {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto description = std::make_unique<cricket::SessionDescription>();
AddAudioSection(description.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(description.get(), kVideoMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
return description;
}
std::unique_ptr<cricket::SessionDescription>
CreateSessionDescriptionWithBundleGroup() {
auto description = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
description->AddGroup(bundle_group);
return description;
}
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
std::unique_ptr<cricket::SessionDescription>
CreateSessionDescriptionWithBundledData() {
auto description = CreateSessionDescriptionWithoutBundle();
AddDataSection(description.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
description->AddGroup(bundle_group);
return description;
}
void AddAudioSection(cricket::SessionDescription* description,
const std::string& mid,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
std::unique_ptr<cricket::AudioContentDescription> audio(
new cricket::AudioContentDescription());
// Set RTCP-mux to be true because the default policy is "mux required".
audio->set_rtcp_mux(true);
description->AddContent(mid, cricket::MediaProtocolType::kRtp,
/*rejected=*/false, std::move(audio));
AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
}
void AddVideoSection(cricket::SessionDescription* description,
const std::string& mid,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
std::unique_ptr<cricket::VideoContentDescription> video(
new cricket::VideoContentDescription());
// Set RTCP-mux to be true because the default policy is "mux required".
video->set_rtcp_mux(true);
description->AddContent(mid, cricket::MediaProtocolType::kRtp,
/*rejected=*/false, std::move(video));
AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
}
void AddDataSection(cricket::SessionDescription* description,
const std::string& mid,
cricket::MediaProtocolType protocol_type,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
RTC_CHECK(protocol_type == cricket::MediaProtocolType::kSctp);
std::unique_ptr<cricket::SctpDataContentDescription> data(
new cricket::SctpDataContentDescription());
data->set_rtcp_mux(true);
description->AddContent(mid, protocol_type,
/*rejected=*/false, std::move(data));
AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
}
void AddTransportInfo(cricket::SessionDescription* description,
const std::string& mid,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
std::unique_ptr<rtc::SSLFingerprint> fingerprint;
if (cert) {
fingerprint = rtc::SSLFingerprint::CreateFromCertificate(*cert);
}
cricket::TransportDescription transport_desc(std::vector<std::string>(),
ufrag, pwd, ice_mode,
conn_role, fingerprint.get());
description->AddTransportInfo(cricket::TransportInfo(mid, transport_desc));
}
cricket::IceConfig CreateIceConfig(
int receiving_timeout,
cricket::ContinualGatheringPolicy continual_gathering_policy) {
cricket::IceConfig config;
config.receiving_timeout = receiving_timeout;
config.continual_gathering_policy = continual_gathering_policy;
return config;
}
Candidate CreateCandidate(const std::string& transport_name, int component) {
Candidate c;
c.set_transport_name(transport_name);
c.set_address(rtc::SocketAddress("192.168.1.1", 8000));
c.set_component(component);
c.set_protocol(cricket::UDP_PROTOCOL_NAME);
c.set_priority(1);
return c;
}
void CreateLocalDescriptionAndCompleteConnectionOnNetworkThread() {
if (!network_thread_->IsCurrent()) {
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
CreateLocalDescriptionAndCompleteConnectionOnNetworkThread();
});
return;
}
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
transport_controller_->MaybeStartGathering();
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
fake_audio_dtls->fake_ice_transport()->SignalCandidateGathered(
fake_audio_dtls->fake_ice_transport(),
CreateCandidate(kAudioMid1, /*component=*/1));
fake_video_dtls->fake_ice_transport()->SignalCandidateGathered(
fake_video_dtls->fake_ice_transport(),
CreateCandidate(kVideoMid1, /*component=*/1));
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(2);
fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
fake_audio_dtls->SetReceiving(true);
fake_video_dtls->SetReceiving(true);
fake_audio_dtls->SetWritable(true);
fake_video_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
}
protected:
void OnConnectionState(cricket::IceConnectionState state) {
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
connection_state_ = state;
++connection_state_signal_count_;
}
void OnStandardizedIceConnectionState(
PeerConnectionInterface::IceConnectionState state) {
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
ice_connection_state_ = state;
++ice_connection_state_signal_count_;
}
void OnCombinedConnectionState(
PeerConnectionInterface::PeerConnectionState state) {
RTC_LOG(LS_INFO) << "OnCombinedConnectionState: "
<< static_cast<int>(state);
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
combined_connection_state_ = state;
++combined_connection_state_signal_count_;
}
void OnGatheringState(cricket::IceGatheringState state) {
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
gathering_state_ = state;
++gathering_state_signal_count_;
}
void OnCandidatesGathered(const std::string& transport_name,
const Candidates& candidates) {
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
candidates_[transport_name].insert(candidates_[transport_name].end(),
candidates.begin(), candidates.end());
++candidates_signal_count_;
}
// JsepTransportController::Observer overrides.
Changes to enable use of DatagramTransport as a data channel transport. PeerConnection now has a new setting in RTCConfiguration to enable use of datagram transport for data channels. There is also a corresponding field trial, which has both a kill-switch and a way to change the default value. PeerConnection's interaction with MediaTransport for data channels has been refactored to work with DataChannelTransportInterface instead. Adds a DataChannelState and OnStateChanged() to the DataChannelSink callbacks. This allows PeerConnection to listen to the data channel's state directly, instead of indirectly by monitoring media transport state. This is necessary to enable use of non-media-transport (eg. datagram transport) data channel transports. For now, PeerConnection watches the state through MediaTransport as well. This will persist until MediaTransport implements the new callback. Datagram transport use is negotiated. As such, an offer that requests to use datagram transport for data channels may be rejected by the answerer. If the offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data channels with an extra x-opaque parameter for datagram transport. If the opaque parameter is rejected (by an answerer without datagram support), the offerer may fall back to SCTP. If DTLS is not enabled, there is no viable fallback. In this case, the data channels are negotiated as media transport data channels. If the receiver does not understand the x-opaque line, it will reject these data channels, and the offerer's data channels will be closed. Bug: webrtc:9719 Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28932}
2019-08-21 10:44:59 -07:00
bool OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
MediaTransportInterface* media_transport,
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b Original change's description: > Reland "Refactor SCTP data channels to use DataChannelTransportInterface." > > Also clears SctpTransport before deleting JsepTransport. > > SctpTransport is ref-counted, but the underlying transport is deleted when > JsepTransport clears the rtp_dtls_transport. This results in crashes when > usrsctp attempts to send outgoing packets through a dangling pointer to the > underlying transport. > > Clearing SctpTransport before DtlsTransport removes the pointer to the > underlying transport before it becomes invalid. > > This fixes a crash in chromium's web platform tests (see > https://chromium-review.googlesource.com/c/chromium/src/+/1776711). > > Original change's description: > > Refactor SCTP data channels to use DataChannelTransportInterface. > > > > This change moves SctpTransport to be owned by JsepTransport, which now > > holds a DataChannelTransport implementation for SCTP when it is used for > > data channels. > > > > This simplifies negotiation and fallback to SCTP. Negotiation can now > > use a composite DataChannelTransport, just as negotiation for RTP uses a > > composite RTP transport. > > > > PeerConnection also has one fewer way it needs to manage data channels. > > It now handles SCTP and datagram- or media-transport-based data channels > > the same way. > > > > There are a few leaky abstractions left. For example, PeerConnection > > calls Start() on the SctpTransport at a particular point in negotiation, > > but does not need to call this for other transports. Similarly, PC > > exposes an interface to the SCTP transport directly to the user; there > > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981 > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29120} Bug: webrtc:9719 Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29290}
2019-09-23 14:53:54 -07:00
DataChannelTransportInterface* data_channel_transport) override {
changed_rtp_transport_by_mid_[mid] = rtp_transport;
if (dtls_transport) {
changed_dtls_transport_by_mid_[mid] = dtls_transport->internal();
} else {
changed_dtls_transport_by_mid_[mid] = nullptr;
}
changed_media_transport_by_mid_[mid] = media_transport;
return true;
}
// Information received from signals from transport controller.
cricket::IceConnectionState connection_state_ =
cricket::kIceConnectionConnecting;
PeerConnectionInterface::IceConnectionState ice_connection_state_ =
PeerConnectionInterface::kIceConnectionNew;
PeerConnectionInterface::PeerConnectionState combined_connection_state_ =
PeerConnectionInterface::PeerConnectionState::kNew;
bool receiving_ = false;
cricket::IceGatheringState gathering_state_ = cricket::kIceGatheringNew;
// transport_name => candidates
std::map<std::string, Candidates> candidates_;
// Counts of each signal emitted.
int connection_state_signal_count_ = 0;
int ice_connection_state_signal_count_ = 0;
int combined_connection_state_signal_count_ = 0;
int receiving_signal_count_ = 0;
int gathering_state_signal_count_ = 0;
int candidates_signal_count_ = 0;
// |network_thread_| should be destroyed after |transport_controller_|
std::unique_ptr<rtc::Thread> network_thread_;
std::unique_ptr<FakeTransportFactory> fake_transport_factory_;
rtc::Thread* const signaling_thread_ = nullptr;
bool signaled_on_non_signaling_thread_ = false;
// Used to verify the SignalRtpTransportChanged/SignalDtlsTransportChanged are
// signaled correctly.
std::map<std::string, RtpTransportInternal*> changed_rtp_transport_by_mid_;
std::map<std::string, cricket::DtlsTransportInternal*>
changed_dtls_transport_by_mid_;
std::map<std::string, MediaTransportInterface*>
changed_media_transport_by_mid_;
// Transport controller needs to be destroyed first, because it may issue
// callbacks that modify the changed_*_by_mid in the destructor.
std::unique_ptr<JsepTransportController> transport_controller_;
};
TEST_F(JsepTransportControllerTest, GetRtpTransport) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto audio_rtp_transport = transport_controller_->GetRtpTransport(kAudioMid1);
auto video_rtp_transport = transport_controller_->GetRtpTransport(kVideoMid1);
EXPECT_NE(nullptr, audio_rtp_transport);
EXPECT_NE(nullptr, video_rtp_transport);
EXPECT_NE(audio_rtp_transport, video_rtp_transport);
// Return nullptr for non-existing ones.
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kAudioMid2));
}
TEST_F(JsepTransportControllerTest, GetDtlsTransport) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_NE(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
EXPECT_NE(nullptr,
transport_controller_->LookupDtlsTransportByMid(kAudioMid1));
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_NE(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
EXPECT_NE(nullptr,
transport_controller_->LookupDtlsTransportByMid(kVideoMid1));
// Lookup for all MIDs should return different transports (no bundle)
EXPECT_NE(transport_controller_->LookupDtlsTransportByMid(kAudioMid1),
transport_controller_->LookupDtlsTransportByMid(kVideoMid1));
// Return nullptr for non-existing ones.
EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kVideoMid2));
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid2));
EXPECT_EQ(nullptr,
transport_controller_->LookupDtlsTransportByMid(kVideoMid2));
// Take a pointer to a transport, shut down the transport controller,
// and verify that the resulting container is empty.
auto dtls_transport =
transport_controller_->LookupDtlsTransportByMid(kVideoMid1);
webrtc::DtlsTransport* my_transport =
static_cast<DtlsTransport*>(dtls_transport.get());
EXPECT_NE(nullptr, my_transport->internal());
transport_controller_.reset();
EXPECT_EQ(nullptr, my_transport->internal());
}
TEST_F(JsepTransportControllerTest, GetDtlsTransportWithRtcpMux) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
}
TEST_F(JsepTransportControllerTest,
DtlsIsStillCreatedIfMediaTransportIsOnlyUsedForDataChannels) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_NE(absl::nullopt,
transport_controller_->GenerateOrGetLastMediaTransportOffer());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
Changes to enable use of DatagramTransport as a data channel transport. PeerConnection now has a new setting in RTCConfiguration to enable use of datagram transport for data channels. There is also a corresponding field trial, which has both a kill-switch and a way to change the default value. PeerConnection's interaction with MediaTransport for data channels has been refactored to work with DataChannelTransportInterface instead. Adds a DataChannelState and OnStateChanged() to the DataChannelSink callbacks. This allows PeerConnection to listen to the data channel's state directly, instead of indirectly by monitoring media transport state. This is necessary to enable use of non-media-transport (eg. datagram transport) data channel transports. For now, PeerConnection watches the state through MediaTransport as well. This will persist until MediaTransport implements the new callback. Datagram transport use is negotiated. As such, an offer that requests to use datagram transport for data channels may be rejected by the answerer. If the offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data channels with an extra x-opaque parameter for datagram transport. If the opaque parameter is rejected (by an answerer without datagram support), the offerer may fall back to SCTP. If DTLS is not enabled, there is no viable fallback. In this case, the data channels are negotiated as media transport data channels. If the receiver does not understand the x-opaque line, it will reject these data channels, and the offerer's data channels will be closed. Bug: webrtc:9719 Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28932}
2019-08-21 10:44:59 -07:00
transport_controller_->GetDataChannelTransport(kAudioMid1));
ASSERT_NE(nullptr, media_transport);
// After SetLocalDescription, media transport should be created as caller.
EXPECT_TRUE(media_transport->is_caller());
EXPECT_TRUE(media_transport->pre_shared_key().has_value());
// Return nullptr for non-existing mids.
EXPECT_EQ(nullptr,
Changes to enable use of DatagramTransport as a data channel transport. PeerConnection now has a new setting in RTCConfiguration to enable use of datagram transport for data channels. There is also a corresponding field trial, which has both a kill-switch and a way to change the default value. PeerConnection's interaction with MediaTransport for data channels has been refactored to work with DataChannelTransportInterface instead. Adds a DataChannelState and OnStateChanged() to the DataChannelSink callbacks. This allows PeerConnection to listen to the data channel's state directly, instead of indirectly by monitoring media transport state. This is necessary to enable use of non-media-transport (eg. datagram transport) data channel transports. For now, PeerConnection watches the state through MediaTransport as well. This will persist until MediaTransport implements the new callback. Datagram transport use is negotiated. As such, an offer that requests to use datagram transport for data channels may be rejected by the answerer. If the offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data channels with an extra x-opaque parameter for datagram transport. If the opaque parameter is rejected (by an answerer without datagram support), the offerer may fall back to SCTP. If DTLS is not enabled, there is no viable fallback. In this case, the data channels are negotiated as media transport data channels. If the receiver does not understand the x-opaque line, it will reject these data channels, and the offerer's data channels will be closed. Bug: webrtc:9719 Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28932}
2019-08-21 10:44:59 -07:00
transport_controller_->GetDataChannelTransport(kVideoMid2));
EXPECT_EQ(cricket::ICE_CANDIDATE_COMPONENT_RTP,
transport_controller_->GetDtlsTransport(kAudioMid1)->component())
<< "Media transport for media was not enabled, and so DTLS transport "
"should be created.";
}
TEST_F(JsepTransportControllerTest,
DtlsIsStillCreatedIfDatagramTransportIsOnlyUsedForDataChannels) {
FakeMediaTransportFactory fake_media_transport_factory("transport_params");
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_datagram_transport_for_data_channels = true;
CreateJsepTransportController(config);
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
auto description = CreateSessionDescriptionWithBundledData();
AddCryptoSettings(description.get());
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
absl::optional<cricket::OpaqueTransportParameters> params =
transport_controller_->GetTransportParameters(kAudioMid1);
for (auto& info : description->transport_infos()) {
info.description.opaque_parameters = params;
}
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
for (cricket::ContentInfo& content_info : description->contents()) {
if (content_info.media_description()->type() == cricket::MEDIA_TYPE_DATA) {
content_info.media_description()->set_alt_protocol(params->protocol);
}
}
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, description.get())
.ok());
FakeDatagramTransport* datagram_transport =
static_cast<FakeDatagramTransport*>(
transport_controller_->GetDataChannelTransport(kAudioMid1));
ASSERT_NE(nullptr, datagram_transport);
EXPECT_EQ(cricket::ICE_CANDIDATE_COMPONENT_RTP,
transport_controller_->GetDtlsTransport(kAudioMid1)->component())
<< "Datagram transport for media was not enabled, and so DTLS transport "
"should be created.";
// Datagram transport is not used for media, so no max packet size is
// specified.
EXPECT_EQ(transport_controller_->GetMediaTransportConfig(kAudioMid1)
.rtp_max_packet_size,
absl::nullopt);
// Since datagram transport is not used for RTP, setting it to writable should
// not make the RTP transport writable.
datagram_transport->set_state(MediaTransportState::kWritable);
EXPECT_FALSE(transport_controller_->GetRtpTransport(kAudioMid1)
->IsWritable(/*rtcp=*/false));
}
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
// An offer that bundles different alt-protocols should be rejected.
TEST_F(JsepTransportControllerTest, CannotBundleDifferentAltProtocols) {
FakeMediaTransportFactory fake_media_transport_factory("transport_params");
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_datagram_transport = true;
config.use_datagram_transport_for_data_channels = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundledData();
AddCryptoSettings(description.get());
absl::optional<cricket::OpaqueTransportParameters> params =
transport_controller_->GetTransportParameters(kAudioMid1);
for (auto& info : description->transport_infos()) {
info.description.opaque_parameters = params;
}
// Append a different alt-protocol to each of the sections.
for (cricket::ContentInfo& content_info : description->contents()) {
content_info.media_description()->set_alt_protocol(params->protocol + "-" +
content_info.name);
}
EXPECT_FALSE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, description.get())
.ok());
}
TEST_F(JsepTransportControllerTest, GetMediaTransportInCaller) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_NE(absl::nullopt,
transport_controller_->GenerateOrGetLastMediaTransportOffer());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
ASSERT_NE(nullptr, media_transport);
// After SetLocalDescription, media transport should be created as caller.
EXPECT_TRUE(media_transport->is_caller());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
// We set the pre-shared key on the caller.
EXPECT_TRUE(media_transport->pre_shared_key().has_value());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_TRUE(media_transport->is_connected());
// Return nullptr for non-existing mids.
EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kVideoMid2));
EXPECT_EQ(cricket::kNoOpDtlsTransportComponent,
transport_controller_->GetDtlsTransport(kAudioMid1)->component())
<< "Because media transport is used, expected no-op DTLS transport.";
}
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
TEST_F(JsepTransportControllerTest,
GetMediaTransportOfferInTheConfigOnSubsequentCalls) {
FakeMediaTransportFactory fake_media_transport_factory;
WrapperMediaTransportFactory wrapping_factory(&fake_media_transport_factory);
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &wrapping_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
transport_controller_->GenerateOrGetLastMediaTransportOffer();
ASSERT_NE(absl::nullopt, settings);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
ASSERT_NE(nullptr, media_transport);
absl::optional<cricket::SessionDescription::MediaTransportSetting>
new_settings =
transport_controller_->GenerateOrGetLastMediaTransportOffer();
ASSERT_NE(absl::nullopt, new_settings);
EXPECT_EQ(settings->transport_name, new_settings->transport_name);
EXPECT_EQ(settings->transport_setting, new_settings->transport_setting);
EXPECT_EQ(1, wrapping_factory.created_transport_count());
}
TEST_F(JsepTransportControllerTest, GetMediaTransportInCallee) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
description->AddMediaTransportSetting("fake", "fake-remote-settings");
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
ASSERT_NE(nullptr, media_transport);
// After SetRemoteDescription, media transport should be created as callee.
EXPECT_FALSE(media_transport->is_caller());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
// We do not set pre-shared key on the callee, it comes in media transport
// settings.
EXPECT_EQ(absl::nullopt, media_transport->settings().pre_shared_key);
EXPECT_TRUE(media_transport->is_connected());
// Return nullptr for non-existing mids.
EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kVideoMid2));
EXPECT_EQ(cricket::kNoOpDtlsTransportComponent,
transport_controller_->GetDtlsTransport(kAudioMid1)->component())
<< "Because media transport is used, expected no-op DTLS transport.";
}
TEST_F(JsepTransportControllerTest, GetMediaTransportInCalleePassesSdp) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
description->AddMediaTransportSetting("fake", "this-is-a-test-setting");
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
ASSERT_NE(nullptr, media_transport);
EXPECT_EQ("this-is-a-test-setting",
media_transport->settings().remote_transport_parameters);
}
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
// Caller generates the offer if media transport returns empty offer (no
// parameters).
TEST_F(JsepTransportControllerTest, MediaTransportGeneratesSessionDescription) {
FakeMediaTransportFactory fake_media_transport_factory(
/*transport_offer=*/"");
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
transport_controller_->GenerateOrGetLastMediaTransportOffer();
ASSERT_TRUE(settings.has_value());
EXPECT_EQ("fake", settings->transport_name);
// Fake media transport returns empty settings (but not nullopt settings!)
EXPECT_EQ("", settings->transport_setting);
}
// Caller generates the offer if media transport returns offer with parameters.
TEST_F(JsepTransportControllerTest,
MediaTransportGeneratesSessionDescriptionWithOfferParams) {
FakeMediaTransportFactory fake_media_transport_factory(
/*transport_offer=*/"offer-params");
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
transport_controller_->GenerateOrGetLastMediaTransportOffer();
ASSERT_TRUE(settings.has_value());
EXPECT_EQ("fake", settings->transport_name);
EXPECT_EQ("offer-params", settings->transport_setting);
}
// Caller skips the offer if media transport requests it.
TEST_F(JsepTransportControllerTest,
MediaTransportGeneratesSkipsSessionDescription) {
FakeMediaTransportFactory fake_media_transport_factory(
/*transport_offer=*/absl::nullopt);
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
transport_controller_->GenerateOrGetLastMediaTransportOffer();
// Fake media transport returns nullopt settings
ASSERT_EQ(absl::nullopt, settings);
}
// Caller ignores its own outgoing parameters.
TEST_F(JsepTransportControllerTest,
GetMediaTransportInCallerIgnoresXmtSection) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_NE(absl::nullopt,
transport_controller_->GenerateOrGetLastMediaTransportOffer());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
ASSERT_NE(nullptr, media_transport);
// Remote parameters are nullopt, because we are the offerer (we don't)
// have the remote transport parameters, only ours.
EXPECT_EQ(absl::nullopt,
media_transport->settings().remote_transport_parameters);
}
TEST_F(JsepTransportControllerTest,
GetMediaTransportInCalleeIgnoresDifferentTransport) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
description->AddMediaTransportSetting("not-a-fake-transport",
"this-is-a-test-setting");
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
ASSERT_NE(nullptr, media_transport);
EXPECT_EQ(absl::nullopt,
media_transport->settings().remote_transport_parameters);
}
TEST_F(JsepTransportControllerTest, GetMediaTransportIsNotSetIfNoSdes) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
// Even if we set local description with crypto now (after the remote offer
// was set), media transport won't be provided.
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
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auto description2 = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description2.get());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, description2.get())
.ok());
EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
EXPECT_EQ(cricket::ICE_CANDIDATE_COMPONENT_RTP,
transport_controller_->GetDtlsTransport(kAudioMid1)->component())
<< "Because media transport is NOT used (fallback to RTP), expected "
"actual DTLS transport for RTP";
}
TEST_F(JsepTransportControllerTest,
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
AfterSettingAnswerTheSameMediaTransportIsReturnedCallee) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
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config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
description->AddMediaTransportSetting("fake", "fake-settings");
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, description.get())
.ok());
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
EXPECT_NE(nullptr, media_transport);
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_FALSE(media_transport->pre_shared_key().has_value())
<< "On the callee, preshared key is passed through the media-transport "
"settings (x-mt)";
// Even if we set local description with crypto now (after the remote offer
// was set), media transport won't be provided.
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
auto description2 = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description2.get());
RTCError result = transport_controller_->SetLocalDescription(
SdpType::kAnswer, description2.get());
EXPECT_TRUE(result.ok()) << result.message();
// Media transport did not change.
EXPECT_EQ(media_transport,
transport_controller_->GetMediaTransport(kAudioMid1));
}
TEST_F(JsepTransportControllerTest, SetIceConfig) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
transport_controller_->SetIceConfig(
CreateIceConfig(kTimeout, cricket::GATHER_CONTINUALLY));
FakeDtlsTransport* fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
ASSERT_NE(nullptr, fake_audio_dtls);
EXPECT_EQ(kTimeout,
fake_audio_dtls->fake_ice_transport()->receiving_timeout());
EXPECT_TRUE(fake_audio_dtls->fake_ice_transport()->gather_continually());
// Test that value stored in controller is applied to new transports.
AddAudioSection(description.get(), kAudioMid2, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid2));
ASSERT_NE(nullptr, fake_audio_dtls);
EXPECT_EQ(kTimeout,
fake_audio_dtls->fake_ice_transport()->receiving_timeout());
EXPECT_TRUE(fake_audio_dtls->fake_ice_transport()->gather_continually());
}
// Tests the getter and setter of the ICE restart flag.
TEST_F(JsepTransportControllerTest, NeedIceRestart) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, description.get())
.ok());
// Initially NeedsIceRestart should return false.
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kAudioMid1));
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kVideoMid1));
// Set the needs-ice-restart flag and verify NeedsIceRestart starts returning
// true.
transport_controller_->SetNeedsIceRestartFlag();
EXPECT_TRUE(transport_controller_->NeedsIceRestart(kAudioMid1));
EXPECT_TRUE(transport_controller_->NeedsIceRestart(kVideoMid1));
// For a nonexistent transport, false should be returned.
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kVideoMid2));
// Reset the ice_ufrag/ice_pwd for audio.
auto audio_transport_info = description->GetTransportInfoByName(kAudioMid1);
audio_transport_info->description.ice_ufrag = kIceUfrag2;
audio_transport_info->description.ice_pwd = kIcePwd2;
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
// Because the ICE is only restarted for audio, NeedsIceRestart is expected to
// return false for audio and true for video.
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kAudioMid1));
EXPECT_TRUE(transport_controller_->NeedsIceRestart(kVideoMid1));
}
TEST_F(JsepTransportControllerTest, MaybeStartGathering) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
// After setting the local description, we should be able to start gathering
// candidates.
transport_controller_->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, AddRemoveRemoteCandidates) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
transport_controller_->SetLocalDescription(SdpType::kOffer,
description.get());
transport_controller_->SetRemoteDescription(SdpType::kAnswer,
description.get());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
ASSERT_NE(nullptr, fake_audio_dtls);
Candidates candidates;
candidates.push_back(
CreateCandidate(kAudioMid1, cricket::ICE_CANDIDATE_COMPONENT_RTP));
EXPECT_TRUE(
transport_controller_->AddRemoteCandidates(kAudioMid1, candidates).ok());
EXPECT_EQ(1U,
fake_audio_dtls->fake_ice_transport()->remote_candidates().size());
EXPECT_TRUE(transport_controller_->RemoveRemoteCandidates(candidates).ok());
EXPECT_EQ(0U,
fake_audio_dtls->fake_ice_transport()->remote_candidates().size());
}
TEST_F(JsepTransportControllerTest, SetAndGetLocalCertificate) {
CreateJsepTransportController(JsepTransportController::Config());
rtc::scoped_refptr<rtc::RTCCertificate> certificate1 =
rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
rtc::scoped_refptr<rtc::RTCCertificate> returned_certificate;
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto description = std::make_unique<cricket::SessionDescription>();
AddAudioSection(description.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
certificate1);
// Apply the local certificate.
EXPECT_TRUE(transport_controller_->SetLocalCertificate(certificate1));
// Apply the local description.
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
returned_certificate = transport_controller_->GetLocalCertificate(kAudioMid1);
EXPECT_TRUE(returned_certificate);
EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
returned_certificate->identity()->certificate().ToPEMString());
// Should fail if called for a nonexistant transport.
EXPECT_EQ(nullptr, transport_controller_->GetLocalCertificate(kVideoMid1));
// Shouldn't be able to change the identity once set.
rtc::scoped_refptr<rtc::RTCCertificate> certificate2 =
rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT)));
EXPECT_FALSE(transport_controller_->SetLocalCertificate(certificate2));
}
TEST_F(JsepTransportControllerTest, GetRemoteSSLCertChain) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
rtc::FakeSSLCertificate fake_certificate("fake_data");
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
fake_audio_dtls->SetRemoteSSLCertificate(&fake_certificate);
std::unique_ptr<rtc::SSLCertChain> returned_cert_chain =
transport_controller_->GetRemoteSSLCertChain(kAudioMid1);
ASSERT_TRUE(returned_cert_chain);
ASSERT_EQ(1u, returned_cert_chain->GetSize());
EXPECT_EQ(fake_certificate.ToPEMString(),
returned_cert_chain->Get(0).ToPEMString());
// Should fail if called for a nonexistant transport.
EXPECT_FALSE(transport_controller_->GetRemoteSSLCertChain(kAudioMid2));
}
TEST_F(JsepTransportControllerTest, GetDtlsRole) {
CreateJsepTransportController(JsepTransportController::Config());
auto offer_certificate =
rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
rtc::SSLIdentity::Generate("offer", rtc::KT_DEFAULT)));
auto answer_certificate =
rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
rtc::SSLIdentity::Generate("answer", rtc::KT_DEFAULT)));
transport_controller_->SetLocalCertificate(offer_certificate);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto offer_desc = std::make_unique<cricket::SessionDescription>();
AddAudioSection(offer_desc.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
offer_certificate);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto answer_desc = std::make_unique<cricket::SessionDescription>();
AddAudioSection(answer_desc.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
answer_certificate);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, offer_desc.get())
.ok());
absl::optional<rtc::SSLRole> role =
transport_controller_->GetDtlsRole(kAudioMid1);
// The DTLS role is not decided yet.
EXPECT_FALSE(role);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, answer_desc.get())
.ok());
role = transport_controller_->GetDtlsRole(kAudioMid1);
ASSERT_TRUE(role);
EXPECT_EQ(rtc::SSL_CLIENT, *role);
}
TEST_F(JsepTransportControllerTest, GetStats) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
cricket::TransportStats stats;
EXPECT_TRUE(transport_controller_->GetStats(kAudioMid1, &stats));
EXPECT_EQ(kAudioMid1, stats.transport_name);
EXPECT_EQ(1u, stats.channel_stats.size());
// Return false for non-existing transport.
EXPECT_FALSE(transport_controller_->GetStats(kAudioMid2, &stats));
}
TEST_F(JsepTransportControllerTest, SignalConnectionStateFailed) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
fake_ice->SetCandidatesGatheringComplete();
fake_ice->SetConnectionCount(1);
// The connection stats will be failed if there is no active connection.
fake_ice->SetConnectionCount(0);
EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
EXPECT_EQ(1, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
ice_connection_state_, kTimeout);
EXPECT_EQ(1, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
combined_connection_state_, kTimeout);
EXPECT_EQ(1, combined_connection_state_signal_count_);
}
TEST_F(JsepTransportControllerTest,
SignalConnectionStateConnectedNoMediaTransport) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
// First, have one transport connect, and another fail, to ensure that
// the first transport connecting didn't trigger a "connected" state signal.
// We should only get a signal when all are connected.
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_audio_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
// Decrease the number of the connection to trigger the signal.
fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_video_dtls->fake_ice_transport()->SetConnectionCount(0);
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
EXPECT_EQ(1, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
ice_connection_state_, kTimeout);
EXPECT_EQ(2, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
combined_connection_state_, kTimeout);
EXPECT_EQ(2, combined_connection_state_signal_count_);
fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
// Set the connection count to be 2 and the cricket::FakeIceTransport will set
// the transport state to be STATE_CONNECTING.
fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
fake_video_dtls->SetWritable(true);
EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
EXPECT_EQ(2, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
ice_connection_state_, kTimeout);
EXPECT_EQ(3, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnected,
combined_connection_state_, kTimeout);
EXPECT_EQ(3, combined_connection_state_signal_count_);
}
TEST_F(JsepTransportControllerTest,
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
SignalConnectionStateConnectedWithMediaTransportAndNoDtlsCaller) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.media_transport_factory = &fake_media_transport_factory;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.use_media_transport_for_data_channels = true;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
// Media Transport is only used with bundle.
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_NE(absl::nullopt,
transport_controller_->GenerateOrGetLastMediaTransportOffer());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
auto fake_video_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1)->ice_transport());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_EQ(fake_audio_ice, fake_video_ice);
fake_audio_ice->SetConnectionCount(2);
fake_audio_ice->SetConnectionCount(1);
fake_video_ice->SetConnectionCount(2);
fake_video_ice->SetConnectionCount(1);
fake_audio_ice->SetWritable(true);
fake_video_ice->SetWritable(true);
// Still not connected, because we are waiting for media transport.
EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
kTimeout);
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
ASSERT_NE(nullptr, media_transport);
media_transport->SetState(webrtc::MediaTransportState::kWritable);
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
// Only one media transport.
EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
}
TEST_F(JsepTransportControllerTest,
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
SignalConnectionStateConnectedWithMediaTransportCaller) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.media_transport_factory = &fake_media_transport_factory;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
// Media Transport is only used with bundle.
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_NE(absl::nullopt,
transport_controller_->GenerateOrGetLastMediaTransportOffer());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
auto fake_audio_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
auto fake_video_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1)->ice_transport());
fake_audio_ice->SetConnectionCount(2);
fake_audio_ice->SetConnectionCount(1);
fake_video_ice->SetConnectionCount(2);
fake_video_ice->SetConnectionCount(1);
fake_audio_ice->SetWritable(true);
fake_video_ice->SetWritable(true);
fake_audio_dtls->SetWritable(true);
fake_video_dtls->SetWritable(true);
// Still not connected, because we are waiting for media transport.
EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
kTimeout);
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
ASSERT_NE(nullptr, media_transport);
media_transport->SetState(webrtc::MediaTransportState::kWritable);
EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
kTimeout);
// Still waiting for the second media transport.
media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kVideoMid1));
media_transport->SetState(webrtc::MediaTransportState::kWritable);
EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
}
TEST_F(JsepTransportControllerTest,
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
SignalConnectionStateFailedWhenMediaTransportClosedCaller) {
FakeMediaTransportFactory fake_media_transport_factory;
JsepTransportController::Config config;
config.media_transport_factory = &fake_media_transport_factory;
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.use_media_transport_for_media = true;
CreateJsepTransportController(config);
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
EXPECT_NE(absl::nullopt,
transport_controller_->GenerateOrGetLastMediaTransportOffer());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
auto fake_audio_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
auto fake_video_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1)->ice_transport());
fake_audio_ice->SetWritable(true);
fake_video_ice->SetWritable(true);
// Decreasing connection count from 2 to 1 triggers connection state event.
fake_audio_ice->SetConnectionCount(2);
fake_audio_ice->SetConnectionCount(1);
fake_video_ice->SetConnectionCount(2);
fake_video_ice->SetConnectionCount(1);
fake_audio_dtls->SetWritable(true);
fake_video_dtls->SetWritable(true);
FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kAudioMid1));
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
ASSERT_NE(nullptr, media_transport);
media_transport->SetState(webrtc::MediaTransportState::kWritable);
media_transport = static_cast<FakeMediaTransport*>(
transport_controller_->GetMediaTransport(kVideoMid1));
Add x-mt line to the offer. We already support decoding of the x-mt line. This change adds the a=x-mt line to the SDP offer. This is not a backward compatible change for media transport (because of the changes in pre-shared key handling) 1) if media transport is enabled, and SDES is enabled, generate the media transport offer. 2) if media transport generated the offer, add that offer to the x-mt line. 3) in order to create media transport, require an x-mt line (backward incompatible). The way it works is that 1) PeerConnection, on the offerer, asks jsep transport for the configuration of the media transport. 2) Tentative media transport is created in JsepTransportController when that happens. 3) SessionDescription will include configuration from this tentative media transport. 4) When the LocalDescription is set on the offerer, the tentative media transport is promoted to the real media transport. Caveats: - now we really only support MaxBundle. In the previous implementations, two media transports were briefly created in some tests, and the second one was destroyed shortly after instantiation. - we, for now, enforce SDES. In the future, whether SDES is used will be refactored out of the peer connection. In the future (on the callee) we should ignore 'is_media_transport' setting. If Offer contains x-mt, media transport should be used (if the factory is present). However, we need to decide how to negotiate media transport for data channels vs data transport for media (x-mt line at this point doesn't differentiate the two, so we still need to use app setting). This change also removes the negotation of pre-shared key from the a=crypto line. Instead, media transport will have its own, 256bit key. Such key should be transported in the x-mt line. This makes the code much simpler, and simplifies the dependency / a=crypto lines parsing. Also, adds a proper test for the connection re-offer (on both sides: callee and caller). Before, it was possible that media transport could get recreated, based on the offer. The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test. This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even when there is a re-offer. Bug: webrtc:9719 Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01 Reviewed-on: https://webrtc-review.googlesource.com/c/125040 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 11:14:05 -08:00
ASSERT_NE(nullptr, media_transport);
media_transport->SetState(webrtc::MediaTransportState::kWritable);
EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
media_transport->SetState(webrtc::MediaTransportState::kClosed);
EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
}
TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
// First, have one transport connect, and another fail, to ensure that
// the first transport connecting didn't trigger a "connected" state signal.
// We should only get a signal when all are connected.
fake_audio_dtls->fake_ice_transport()->SetTransportState(
IceTransportState::kCompleted,
cricket::IceTransportState::STATE_COMPLETED);
fake_audio_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
ice_connection_state_, kTimeout);
EXPECT_EQ(1, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnecting,
combined_connection_state_, kTimeout);
EXPECT_EQ(1, combined_connection_state_signal_count_);
fake_video_dtls->fake_ice_transport()->SetTransportState(
IceTransportState::kFailed, cricket::IceTransportState::STATE_FAILED);
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
EXPECT_EQ(1, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
ice_connection_state_, kTimeout);
EXPECT_EQ(2, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
combined_connection_state_, kTimeout);
EXPECT_EQ(2, combined_connection_state_signal_count_);
fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
// Set the connection count to be 1 and the cricket::FakeIceTransport will set
// the transport state to be STATE_COMPLETED.
fake_video_dtls->fake_ice_transport()->SetTransportState(
IceTransportState::kCompleted,
cricket::IceTransportState::STATE_COMPLETED);
fake_video_dtls->SetWritable(true);
EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
EXPECT_EQ(3, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
ice_connection_state_, kTimeout);
EXPECT_EQ(3, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnected,
combined_connection_state_, kTimeout);
EXPECT_EQ(3, combined_connection_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, SignalIceGatheringStateGathering) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
// Should be in the gathering state as soon as any transport starts gathering.
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, SignalIceGatheringStateComplete) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
// Have one transport finish gathering, to make sure gathering
// completion wasn't signalled if only one transport finished gathering.
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ(1, gathering_state_signal_count_);
fake_video_dtls->fake_ice_transport()->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
EXPECT_EQ(2, gathering_state_signal_count_);
}
// Test that when the last transport that hasn't finished connecting and/or
// gathering is destroyed, the aggregate state jumps to "completed". This can
// happen if, for example, we have an audio and video transport, the audio
// transport completes, then we start bundling video on the audio transport.
TEST_F(JsepTransportControllerTest,
SignalingWhenLastIncompleteTransportDestroyed) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_NE(fake_audio_dtls, fake_video_dtls);
fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
// Let the audio transport complete.
fake_audio_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
EXPECT_EQ(1, gathering_state_signal_count_);
// Set the remote description and enable the bundle.
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, description.get())
.ok());
// The BUNDLE should be enabled, the incomplete video transport should be
// deleted and the states shoud be updated.
fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_EQ(fake_audio_dtls, fake_video_dtls);
EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
EXPECT_EQ(PeerConnectionInterface::kIceConnectionCompleted,
ice_connection_state_);
EXPECT_EQ(PeerConnectionInterface::PeerConnectionState::kConnected,
combined_connection_state_);
EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
EXPECT_EQ(2, gathering_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, SignalCandidatesGathered) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
transport_controller_->MaybeStartGathering();
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
fake_audio_dtls->fake_ice_transport()->SignalCandidateGathered(
fake_audio_dtls->fake_ice_transport(), CreateCandidate(kAudioMid1, 1));
EXPECT_EQ_WAIT(1, candidates_signal_count_, kTimeout);
EXPECT_EQ(1u, candidates_[kAudioMid1].size());
}
TEST_F(JsepTransportControllerTest, IceSignalingOccursOnSignalingThread) {
network_thread_ = rtc::Thread::CreateWithSocketServer();
network_thread_->Start();
CreateJsepTransportController(JsepTransportController::Config(),
signaling_thread_, network_thread_.get(),
/*port_allocator=*/nullptr);
CreateLocalDescriptionAndCompleteConnectionOnNetworkThread();
// connecting --> connected --> completed
EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
EXPECT_EQ(2, connection_state_signal_count_);
// new --> gathering --> complete
EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
EXPECT_EQ(2, gathering_state_signal_count_);
EXPECT_EQ_WAIT(1u, candidates_[kAudioMid1].size(), kTimeout);
EXPECT_EQ_WAIT(1u, candidates_[kVideoMid1].size(), kTimeout);
EXPECT_EQ(2, candidates_signal_count_);
EXPECT_TRUE(!signaled_on_non_signaling_thread_);
}
// Older versions of Chrome expect the ICE role to be re-determined when an
// ICE restart occurs, and also don't perform conflict resolution correctly,
// so for now we can't safely stop doing this.
// See: https://bugs.chromium.org/p/chromium/issues/detail?id=628676
// TODO(deadbeef): Remove this when these old versions of Chrome reach a low
// enough population.
TEST_F(JsepTransportControllerTest, IceRoleRedeterminedOnIceRestartByDefault) {
CreateJsepTransportController(JsepTransportController::Config());
// Let the |transport_controller_| be the controlled side initially.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, local_answer.get())
.ok());
auto fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
fake_dtls->fake_ice_transport()->GetIceRole());
// New offer will trigger the ICE restart.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto restart_local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(restart_local_offer.get(), kAudioMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(
transport_controller_
->SetLocalDescription(SdpType::kOffer, restart_local_offer.get())
.ok());
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
}
// Test that if the TransportController was created with the
// |redetermine_role_on_ice_restart| parameter set to false, the role is *not*
// redetermined on an ICE restart.
TEST_F(JsepTransportControllerTest, IceRoleNotRedetermined) {
JsepTransportController::Config config;
config.redetermine_role_on_ice_restart = false;
CreateJsepTransportController(config);
// Let the |transport_controller_| be the controlled side initially.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, local_answer.get())
.ok());
auto fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
fake_dtls->fake_ice_transport()->GetIceRole());
// New offer will trigger the ICE restart.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto restart_local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(restart_local_offer.get(), kAudioMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(
transport_controller_
->SetLocalDescription(SdpType::kOffer, restart_local_offer.get())
.ok());
EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
fake_dtls->fake_ice_transport()->GetIceRole());
}
// Tests ICE-Lite mode in remote answer.
TEST_F(JsepTransportControllerTest, SetIceRoleWhenIceLiteInRemoteAnswer) {
CreateJsepTransportController(JsepTransportController::Config());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
auto fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
EXPECT_EQ(cricket::ICEMODE_FULL,
fake_dtls->fake_ice_transport()->remote_ice_mode());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
EXPECT_EQ(cricket::ICEMODE_LITE,
fake_dtls->fake_ice_transport()->remote_ice_mode());
}
// Tests that the ICE role remains "controlling" if a subsequent offer that
// does an ICE restart is received from an ICE lite endpoint. Regression test
// for: https://crbug.com/710760
TEST_F(JsepTransportControllerTest,
IceRoleIsControllingAfterIceRestartFromIceLiteEndpoint) {
CreateJsepTransportController(JsepTransportController::Config());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Initial Offer/Answer exchange. If the remote offerer is ICE-Lite, then the
// local side is the controlling.
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, local_answer.get())
.ok());
auto fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
// In the subsequence remote offer triggers an ICE restart.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_offer2 = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_offer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_answer2 = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_answer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, remote_offer2.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, local_answer2.get())
.ok());
fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
// The local side is still the controlling role since the remote side is using
// ICE-Lite.
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
}
// Tests that the SDP has more than one audio/video m= sections.
TEST_F(JsepTransportControllerTest, MultipleMediaSectionsOfSameTypeWithBundle) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kAudioMid2);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddAudioSection(local_offer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddAudioSection(remote_answer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Verify that all the sections are bundled on kAudio1.
auto transport1 = transport_controller_->GetRtpTransport(kAudioMid1);
auto transport2 = transport_controller_->GetRtpTransport(kAudioMid2);
auto transport3 = transport_controller_->GetRtpTransport(kVideoMid1);
auto transport4 = transport_controller_->GetRtpTransport(kDataMid1);
EXPECT_EQ(transport1, transport2);
EXPECT_EQ(transport1, transport3);
EXPECT_EQ(transport1, transport4);
EXPECT_EQ(transport_controller_->LookupDtlsTransportByMid(kAudioMid1),
transport_controller_->LookupDtlsTransportByMid(kVideoMid1));
// Verify the OnRtpTransport/DtlsTransportChanged signals are fired correctly.
auto it = changed_rtp_transport_by_mid_.find(kAudioMid2);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
it = changed_rtp_transport_by_mid_.find(kAudioMid2);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
it = changed_rtp_transport_by_mid_.find(kVideoMid1);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
// Verify the DtlsTransport for the SCTP data channel is reset correctly.
auto it2 = changed_dtls_transport_by_mid_.find(kDataMid1);
ASSERT_TRUE(it2 != changed_dtls_transport_by_mid_.end());
}
// Tests that only a subset of all the m= sections are bundled.
TEST_F(JsepTransportControllerTest, BundleSubsetOfMediaSections) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddAudioSection(local_offer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddAudioSection(remote_answer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Verifiy that only |kAudio1| and |kVideo1| are bundled.
auto transport1 = transport_controller_->GetRtpTransport(kAudioMid1);
auto transport2 = transport_controller_->GetRtpTransport(kAudioMid2);
auto transport3 = transport_controller_->GetRtpTransport(kVideoMid1);
EXPECT_NE(transport1, transport2);
EXPECT_EQ(transport1, transport3);
auto it = changed_rtp_transport_by_mid_.find(kVideoMid1);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
it = changed_rtp_transport_by_mid_.find(kAudioMid2);
EXPECT_TRUE(transport2 == it->second);
}
// Tests that the initial offer/answer only have data section and audio/video
// sections are added in the subsequent offer.
TEST_F(JsepTransportControllerTest, BundleOnDataSectionInSubsequentOffer) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
auto data_transport = transport_controller_->GetRtpTransport(kDataMid1);
// Add audio/video sections in subsequent offer.
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Reset the bundle group and do another offer/answer exchange.
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
local_offer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
auto audio_transport = transport_controller_->GetRtpTransport(kAudioMid1);
auto video_transport = transport_controller_->GetRtpTransport(kVideoMid1);
EXPECT_EQ(data_transport, audio_transport);
EXPECT_EQ(data_transport, video_transport);
}
TEST_F(JsepTransportControllerTest, VideoDataRejectedInAnswer) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Reject video and data section.
remote_answer->contents()[1].rejected = true;
remote_answer->contents()[2].rejected = true;
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Verify the RtpTransport/DtlsTransport is destroyed correctly.
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kDataMid1));
// Verify the signals are fired correctly.
auto it = changed_rtp_transport_by_mid_.find(kVideoMid1);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(nullptr, it->second);
auto it2 = changed_dtls_transport_by_mid_.find(kDataMid1);
ASSERT_TRUE(it2 != changed_dtls_transport_by_mid_.end());
EXPECT_EQ(nullptr, it2->second);
}
// Tests that changing the bundled MID in subsequent offer/answer exchange is
// not supported.
// TODO(bugs.webrtc.org/6704): Change this test to expect success once issue is
// fixed
TEST_F(JsepTransportControllerTest, ChangeBundledMidNotSupported) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
EXPECT_EQ(transport_controller_->GetRtpTransport(kAudioMid1),
transport_controller_->GetRtpTransport(kVideoMid1));
// Reorder the bundle group.
EXPECT_TRUE(bundle_group.RemoveContentName(kAudioMid1));
bundle_group.AddContentName(kAudioMid1);
// The answerer uses the new bundle group and now the bundle mid is changed to
// |kVideo1|.
remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// Test that rejecting only the first m= section of a BUNDLE group is treated as
// an error, but rejecting all of them works as expected.
TEST_F(JsepTransportControllerTest, RejectFirstContentInBundleGroup) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Reject audio content in answer.
remote_answer->contents()[0].rejected = true;
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Reject all the contents.
remote_answer->contents()[1].rejected = true;
remote_answer->contents()[2].rejected = true;
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kAudioMid1));
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kDataMid1));
}
// Tests that applying non-RTCP-mux offer would fail when kRtcpMuxPolicyRequire
// is used.
TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxOfferWhenMuxingRequired) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
CreateJsepTransportController(config);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
local_offer->contents()[0].media_description()->set_rtcp_mux(false);
// Applying a non-RTCP-mux offer is expected to fail.
EXPECT_FALSE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
}
// Tests that applying non-RTCP-mux answer would fail when kRtcpMuxPolicyRequire
// is used.
TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxAnswerWhenMuxingRequired) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
CreateJsepTransportController(config);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Applying a non-RTCP-mux answer is expected to fail.
remote_answer->contents()[0].media_description()->set_rtcp_mux(false);
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// This tests that the BUNDLE group in answer should be a subset of the offered
// group.
TEST_F(JsepTransportControllerTest,
AddContentToBundleGroupInAnswerNotSupported) {
CreateJsepTransportController(JsepTransportController::Config());
auto local_offer = CreateSessionDescriptionWithoutBundle();
auto remote_answer = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup offer_bundle_group(cricket::GROUP_TYPE_BUNDLE);
offer_bundle_group.AddContentName(kAudioMid1);
local_offer->AddGroup(offer_bundle_group);
cricket::ContentGroup answer_bundle_group(cricket::GROUP_TYPE_BUNDLE);
answer_bundle_group.AddContentName(kAudioMid1);
answer_bundle_group.AddContentName(kVideoMid1);
remote_answer->AddGroup(answer_bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// This tests that the BUNDLE group with non-existing MID should be rejectd.
TEST_F(JsepTransportControllerTest, RejectBundleGroupWithNonExistingMid) {
CreateJsepTransportController(JsepTransportController::Config());
auto local_offer = CreateSessionDescriptionWithoutBundle();
auto remote_answer = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup invalid_bundle_group(cricket::GROUP_TYPE_BUNDLE);
// The BUNDLE group is invalid because there is no data section in the
// description.
invalid_bundle_group.AddContentName(kDataMid1);
local_offer->AddGroup(invalid_bundle_group);
remote_answer->AddGroup(invalid_bundle_group);
EXPECT_FALSE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// This tests that an answer shouldn't be able to remove an m= section from an
// established group without rejecting it.
TEST_F(JsepTransportControllerTest, RemoveContentFromBundleGroup) {
CreateJsepTransportController(JsepTransportController::Config());
auto local_offer = CreateSessionDescriptionWithBundleGroup();
auto remote_answer = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Do an re-offer/answer.
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
auto new_answer = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
// The answer removes video from the BUNDLE group without rejecting it is
// invalid.
new_bundle_group.AddContentName(kAudioMid1);
new_answer->AddGroup(new_bundle_group);
// Applying invalid answer is expected to fail.
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
.ok());
// Rejected the video content.
auto video_content = new_answer->GetContentByName(kVideoMid1);
ASSERT_TRUE(video_content);
video_content->rejected = true;
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
.ok());
}
// Test that the JsepTransportController can process a new local and remote
// description that changes the tagged BUNDLE group with the max-bundle policy
// specified.
// This is a regression test for bugs.webrtc.org/9954
TEST_F(JsepTransportControllerTest, ChangeTaggedMediaSectionMaxBundle) {
CreateJsepTransportController(JsepTransportController::Config());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
local_offer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
std::unique_ptr<cricket::SessionDescription> remote_answer(
local_offer->Clone());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
std::unique_ptr<cricket::SessionDescription> local_reoffer(
local_offer->Clone());
local_reoffer->contents()[0].rejected = true;
AddVideoSection(local_reoffer.get(), kVideoMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
local_reoffer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
new_bundle_group.AddContentName(kVideoMid1);
local_reoffer->AddGroup(new_bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_reoffer.get())
.ok());
std::unique_ptr<cricket::SessionDescription> remote_reanswer(
local_reoffer->Clone());
EXPECT_TRUE(
transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_reanswer.get())
.ok());
}
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
constexpr char kFakeTransportParameters[] = "fake-params";
// Test fixture that provides common setup and helpers for tests related to the
// datagram transport.
class JsepTransportControllerDatagramTest
: public JsepTransportControllerTest,
public testing::WithParamInterface<bool> {
public:
JsepTransportControllerDatagramTest()
: JsepTransportControllerTest(),
fake_media_transport_factory_(kFakeTransportParameters) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.media_transport_factory = &fake_media_transport_factory_;
config.use_datagram_transport = true;
CreateJsepTransportController(config);
}
// Whether the JsepTransportController under test acts as the offerer or
// answerer in this test.
bool IsOfferer() { return GetParam(); }
// Sets a description as local or remote based on type and current
// perspective.
RTCError SetDescription(SdpType type,
const cricket::SessionDescription* description) {
if (IsOfferer() == (type == SdpType::kOffer)) {
return transport_controller_->SetLocalDescription(type, description);
} else {
return transport_controller_->SetRemoteDescription(type, description);
}
}
// Creates a session description with the settings necessary for datagram
// transport (bundle + crypto) and the given |transport_params|.
std::unique_ptr<cricket::SessionDescription>
CreateSessionDescriptionForDatagramTransport(
absl::optional<cricket::OpaqueTransportParameters> transport_params) {
auto description = CreateSessionDescriptionWithBundleGroup();
AddCryptoSettings(description.get());
for (auto& info : description->transport_infos()) {
info.description.opaque_parameters = transport_params;
}
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
if (transport_params) {
for (auto& content_info : description->contents()) {
content_info.media_description()->set_alt_protocol(
transport_params->protocol);
}
}
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
return description;
}
// Creates transport parameters with |protocol| and |parameters|
// matching what |fake_media_transport_factory_| provides.
cricket::OpaqueTransportParameters CreateTransportParameters() {
cricket::OpaqueTransportParameters params;
params.protocol = fake_media_transport_factory_.GetTransportName();
params.parameters = "fake-params";
return params;
}
protected:
FakeMediaTransportFactory fake_media_transport_factory_;
};
TEST_P(JsepTransportControllerDatagramTest, InitDatagramTransport) {
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
if (IsOfferer()) {
// Getting transport parameters is allowed before setting a description.
// This is necessary so that the offerer can include these params.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
// Setting a description activates the datagram transport without changing
// transport parameters.
auto description = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, description.get()).ok());
// After setting an offer with transport parameters, those parameters are
// reflected by the controller.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
TEST_P(JsepTransportControllerDatagramTest,
OfferMissingDatagramTransportParams) {
if (IsOfferer()) {
// This test doesn't make sense from the offerer's perspective, as the offer
// must contain datagram transport params if the offerer supports it.
return;
}
auto description =
CreateSessionDescriptionForDatagramTransport(absl::nullopt);
EXPECT_TRUE(SetDescription(SdpType::kOffer, description.get()).ok());
// The offer didn't contain any datagram transport parameters, so the answer
// won't either.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
absl::nullopt);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
absl::nullopt);
}
TEST_P(JsepTransportControllerDatagramTest, OfferHasWrongTransportName) {
if (IsOfferer()) {
// This test doesn't make sense from the offerer's perspective, as the
// offerer cannot offer itself the wrong transport.
return;
}
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
fake_params.protocol = "wrong-name";
auto description = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, description.get()).ok());
// The offerer and answerer support different datagram transports, so the
// answerer rejects the offered parameters.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
absl::nullopt);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
absl::nullopt);
}
TEST_P(JsepTransportControllerDatagramTest, AnswerRejectsDatagram) {
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
if (IsOfferer()) {
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
auto offer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, offer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
auto answer = CreateSessionDescriptionForDatagramTransport(absl::nullopt);
EXPECT_TRUE(SetDescription(SdpType::kAnswer, answer.get()).ok());
// The answer rejected datagram transport, so its parameters are empty.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
absl::nullopt);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
absl::nullopt);
}
TEST_P(JsepTransportControllerDatagramTest, AnswerAcceptsDatagram) {
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
if (IsOfferer()) {
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
auto offer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, offer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
auto answer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kAnswer, answer.get()).ok());
// The answer accepted datagram transport, so it is present.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
TEST_P(JsepTransportControllerDatagramTest, PrAnswerRejectsDatagram) {
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
if (IsOfferer()) {
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
auto offer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, offer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
auto answer = CreateSessionDescriptionForDatagramTransport(absl::nullopt);
EXPECT_TRUE(SetDescription(SdpType::kPrAnswer, answer.get()).ok());
// The answer rejected datagram transport, but it's provisional, so the
// transport is kept around for now.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
TEST_P(JsepTransportControllerDatagramTest, PrAnswerAcceptsDatagram) {
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
if (IsOfferer()) {
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
auto offer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, offer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
auto answer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kPrAnswer, answer.get()).ok());
// The answer provisionally accepted datagram transport, so it's kept.
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
TEST_P(JsepTransportControllerDatagramTest, RenegotiationCannotAddDatagram) {
auto offer = CreateSessionDescriptionForDatagramTransport(absl::nullopt);
EXPECT_TRUE(SetDescription(SdpType::kOffer, offer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
absl::nullopt);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
absl::nullopt);
auto answer = CreateSessionDescriptionForDatagramTransport(absl::nullopt);
EXPECT_TRUE(SetDescription(SdpType::kAnswer, answer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
absl::nullopt);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
absl::nullopt);
// Attempting to add a datagram transport on a re-offer does not cause an
// error, but also does not add a datagram transport.
auto reoffer =
CreateSessionDescriptionForDatagramTransport(CreateTransportParameters());
EXPECT_TRUE(SetDescription(SdpType::kOffer, reoffer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
absl::nullopt);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
absl::nullopt);
}
TEST_P(JsepTransportControllerDatagramTest, RenegotiationCannotRemoveDatagram) {
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
if (IsOfferer()) {
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
auto offer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, offer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
auto answer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kAnswer, answer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
// Attempting to remove a datagram transport on a re-offer does not cause an
// error, but also does not remove the datagram transport.
auto reoffer = CreateSessionDescriptionForDatagramTransport(absl::nullopt);
EXPECT_TRUE(SetDescription(SdpType::kOffer, reoffer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
TEST_P(JsepTransportControllerDatagramTest,
RenegotiationKeepsDatagramTransport) {
cricket::OpaqueTransportParameters fake_params = CreateTransportParameters();
if (IsOfferer()) {
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
auto offer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, offer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
auto answer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kAnswer, answer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
// Attempting to remove a datagram transport on a re-offer does not cause an
// error, but also does not remove the datagram transport.
auto reoffer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kOffer, reoffer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
auto reanswer = CreateSessionDescriptionForDatagramTransport(fake_params);
EXPECT_TRUE(SetDescription(SdpType::kAnswer, reanswer.get()).ok());
EXPECT_EQ(transport_controller_->GetTransportParameters(kAudioMid1),
fake_params);
EXPECT_EQ(transport_controller_->GetTransportParameters(kVideoMid1),
fake_params);
}
INSTANTIATE_TEST_SUITE_P(
JsepTransportControllerDatagramTests,
JsepTransportControllerDatagramTest,
testing::Values(true, false),
// The parameter value is the local perspective (offerer or answerer).
[](const testing::TestParamInfo<bool>& info) {
return info.param ? "Offerer" : "Answerer";
});
} // namespace webrtc