59 Commits

Author SHA1 Message Date
Honghai Zhang
161a586b45 Fix some flaky tests by using longer timeout and/or fake clock.
Also use const variables for timeout values.

BUG=webrtc:6500
R=deadbeef@webrtc.org

Review URL: https://codereview.webrtc.org/2431473004 .

Cr-Commit-Position: refs/heads/master@{#14711}
2016-10-20 18:47:19 +00:00
Honghai Zhang
a74363c998 Remove ports that are not used by any channel after timeout
If a port is not used by any channel and if it has no connection for 30
seconds, it will be removed.
Note, as long as a port is used by a transport channel, it will be kept
even if it does not have any connection. This will be beneficial to
continual gathering because new connections can be created in the future
when network changes.

BUG=
R=pthatcher@webrtc.org, zhihuang@webrtc.org

Review URL: https://codereview.webrtc.org/2171183002 .

Cr-Commit-Position: refs/heads/master@{#13567}
2016-07-29 01:06:26 +00:00
skvlad
c309e0e3ea Don't stop sending media on EWOULDBLOCK
This change makes WebRTC no longer stop sending video when we receive an
EWOULDBLOCK error from the operating system. This was previously
causing calls on a slow link (where the first hop is slow) to rapidly
oscillate between starting and stopping video.

We still do need to stop sending packets if there is no known good
connection we can use for that. We used to generate a synthetic
EWOULDBLOCK error in that case. This CL replaces it with a different
code (ENOTCONN); EWOULDBLOCK no longer stops the stream but ENOTCONN
does.

I've updated all the places where we seemed to be generating EWOULDBLOCK
for reasons other than some buffer been full; please give it a thorough
look in case I missed something.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2192963002 .

Cr-Commit-Position: refs/heads/master@{#13566}
2016-07-29 00:15:30 +00:00
Honghai Zhang
b5db1ec0e5 Delay destroying a port if new connections are created and destroyed.
If all connections on a port is destroyed, it will schedule an event
to check if it is dead after a timeout. Previously if a new connection
is created but destroyed before the event is fired, it will destroy the
port. With this change, we will not destoy it until it times out again
after the last created connection is destroyed.

BUG=
R=pthatcher@webrtc.org, zhihuang@webrtc.org

Review URL: https://codereview.webrtc.org/2184013003 .

Cr-Commit-Position: refs/heads/master@{#13563}
2016-07-28 20:23:13 +00:00
Honghai Zhang
b9e7b4ad66 Add config to prune low-priority TURN ports for creating connections
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).

This effectively reduces the number of TURN candidates and connections created by TURN ports.

BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2093623004 .

Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
Cr-Original-Commit-Position: refs/heads/master@{#13335}
Cr-Commit-Position: refs/heads/master@{#13354}
2016-07-01 03:52:16 +00:00
danilchap
f4e8cf0d5b Revert of Add config to prune TURN ports (patchset #12 id:360001 of https://codereview.webrtc.org/2093623004/ )
Reason for revert:
Breaks Win32/Win64 Debug bots in client.webrtc waterfall

Original issue's description:
> Add config to prune low-priority TURN ports for creating connections
> When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
>
> This effectively reduces the number of TURN candidates and connections created by TURN ports.
>
> BUG=
> R=deadbeef@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
> Cr-Commit-Position: refs/heads/master@{#13335}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2111663003
Cr-Commit-Position: refs/heads/master@{#13342}
2016-06-30 08:55:10 +00:00
Honghai Zhang
17aac053f5 Add config to prune low-priority TURN ports for creating connections
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).

This effectively reduces the number of TURN candidates and connections created by TURN ports.

BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2093623004 .

Cr-Commit-Position: refs/heads/master@{#13335}
2016-06-30 04:42:05 +00:00
Taylor Brandstetter
6bb1ef2b86 Fixing bug where Connection drops packets when presumed writable.
The "should I simulate EWOULDBLOCK?" determination now happens
solely in P2PTransportChannel. This also fixes a bug where the
"last packet id" was set even if no packet was sent.

R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2099783002 .

Cr-Commit-Position: refs/heads/master@{#13307}
2016-06-28 01:09:10 +00:00
zhihuang
5ecf16c072 Add Stats to Stun ping.
Add sent_ping_requests, recv_ping_responses to ConnectionInfo.
recv_ping_responses_ will be incremented when OnConnectionRequestResponse() is called.
ent_ping_requests_ will be incremented when OnConnectionRequestSent() is called.

BUG=webrtc:5695

Review-Url: https://codereview.webrtc.org/1940493002
Cr-Commit-Position: refs/heads/master@{#13001}
2016-06-02 00:09:24 +00:00
honghaiz
36f50e8e4e Create a new connection if a candidate reuses an address
If the remote side sends a candidate with the same address and port with an existing candidate,
but with a new ufrag and pwd, the local client will create a new connection from it
and destroy the old connection with the same remote address.

BUG=webrtc:5915

Review-Url: https://codereview.webrtc.org/2018693002
Cr-Commit-Position: refs/heads/master@{#13000}
2016-06-01 22:57:12 +00:00
zhihuang
6d0d4bf31d Change the size of the ICE ufrag to 4 bytes.
This is the minumum allowed size, and will allow STUN pings to be smaller.
The unit tests on the the Gturn are also modified. A username with length of 16 bytes will be generated for Gturn only.

Review-Url: https://codereview.webrtc.org/1848083002
Cr-Commit-Position: refs/heads/master@{#12876}
2016-05-24 17:13:41 +00:00
Honghai Zhang
351d77b702 Update the type and cost of existing networks
if the network monitor detects it after the native code does.

Also set the network cost for ethernet, wifi, unknown, cellular network type to be 0, 10, 50, 900,
so that unknown networks will have lower precedence than known networks with low cost (like Wifi) but  higher precedence than known networks with high cost.

And third, infer network type based on limited name matching in Android if there is no network monitor or network monitor did not find the type.

BUG=webrtc:5890
R=pthatcher@chromium.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1976683003 .

Cr-Commit-Position: refs/heads/master@{#12833}
2016-05-20 22:08:37 +00:00
Taylor Brandstetter
a1c303535f Relanding: Implement RTCConfiguration.iceCandidatePoolSize.
Depends on this CL in order to work in Chromium:
https://codereview.chromium.org/1976673002/

It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).

This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.

R=pthatcher@webrtc.org

Committed: 48e9d05f51

Review URL: https://codereview.webrtc.org/1956453003 .

Cr-Commit-Position: refs/heads/master@{#12729}
2016-05-13 15:15:20 +00:00
deadbeef
c55fb30649 Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ )
Reason for revert:
Breaks remoting_unittests. They defined their own operator== which conflicts with this one.

I'll remove the operator== in a roll CL. But until it's approved, I'm reverting this so the FYI bots will pass.

Original issue's description:
> Implement RTCConfiguration.iceCandidatePoolSize.
>
> It works by creating pooled PortAllocatorSessions which can be picked up
> by a P2PTransportChannel when needed (after a local description is set).
>
> This can optimize candidate gathering time when there is some time between
> creating a PeerConnection and setting a local description.
>
> R=pthatcher@webrtc.org
>
> Committed: 48e9d05f51

TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/1972043004
Cr-Commit-Position: refs/heads/master@{#12709}
2016-05-12 19:51:45 +00:00
Taylor Brandstetter
48e9d05f51 Implement RTCConfiguration.iceCandidatePoolSize.
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).

This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1956453003 .

Cr-Commit-Position: refs/heads/master@{#12708}
2016-05-12 17:19:44 +00:00
nisse
1bffc1d1a4 Rename rtc::Time64 --> rtc::TimeMillis.
In the discussion on https://codereview.webrtc.org/1888593004/, a more
decriptive name was suggested for Time64.

BUG=webrtc:5740

Review-Url: https://codereview.webrtc.org/1923213002
Cr-Commit-Position: refs/heads/master@{#12594}
2016-05-02 15:19:00 +00:00
nisse
ef8b61e110 Enable -Winconsistent-missing-override flag.
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.

NOPRESUBMIT=True
BUG=webrtc:3970

Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
2016-04-29 13:09:23 +00:00
kwiberg
3ec4679dd2 Replace scoped_ptr with unique_ptr in webrtc/p2p/
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1923163003

Cr-Commit-Position: refs/heads/master@{#12532}
2016-04-27 14:22:58 +00:00
jbauch
f1f87203d7 Split ByteBuffer into writer/reader objects.
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.

BUG=webrtc:5155,webrtc:5670

Review URL: https://codereview.webrtc.org/1821083002

Cr-Commit-Position: refs/heads/master@{#12160}
2016-03-30 13:43:44 +00:00
honghaiz
a0c44eaa82 Add 16-bit network id to the candidate signaling.
Also include that in the stun-ping request as part of the
network-info attribute.
Change the network cost to be 16 bits.

BUG=

Review URL: https://codereview.webrtc.org/1815473002

Cr-Commit-Position: refs/heads/master@{#12110}
2016-03-23 23:07:54 +00:00
honghaiz
34b11eb66e Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.

The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.

BUG=webrtc:5636

Review URL: https://codereview.webrtc.org/1793553002

Cr-Commit-Position: refs/heads/master@{#12019}
2016-03-16 15:55:48 +00:00
kwiberg
6baec0351a Port::GetStunMessage: Write to scoped_ptr instead of raw pointer
This is a good idea in general, because it makes ownership clearer,
but will also be very convenient when scoped_ptr is gone, since
unique_ptr doesn't have an .accept() method.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1800753003

Cr-Commit-Position: refs/heads/master@{#12002}
2016-03-15 18:09:59 +00:00
Honghai Zhang
80f1db971d Include relay protocol type when computing the turn candidate foundation.
BUG=576353
R=deadbeef@webrtc.org, pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1619213003 .

Cr-Commit-Position: refs/heads/master@{#11400}
2016-01-27 19:54:44 +00:00
Stefan Holmer
55674ffb32 Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.

R=tommi@webrtc.org
TBR=pthatcher@webtrc.org

BUG=4173

Review URL: https://codereview.webrtc.org/1589563003 .

Cr-Commit-Position: refs/heads/master@{#11251}
2016-01-14 14:49:23 +00:00
tommi
e5e0e57bdf Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
Reason for revert:
Broke Chrome:

https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_chromeos_compile_dbg_ng/builds/143025/steps/compile%20%28with%20patch%29/logs/stdio

FAILED: cd ../../third_party/libjingle; python ../../native_client/build/build_nexe.py --root ../.. --product-dir ../../out/Debug/xyz --config-name Debug -t ../../native_client/toolchain/ --arch pnacl --build newlib_plib --name ../../out/Debug/gen/tc_pnacl_newlib/lib/libjingle_nacl.a --objdir ../../out/Debug/obj/third_party/libjingle/libjingle_nacl.gen/pnacl_newlib-pnacl/libjingle_nacl "--include-dirs=../../out/Debug/gen/tc_pnacl_newlib/include ../.. \"../../out/Debug/gen\" ./source ../ ../../native_client_sdk/src/libraries ../../native_client_sdk/src/libraries/nacl_io/include ../../native_client_sdk/src/libraries/third_party/newlib-extras ../expat/files/lib ../boringssl/src/include" "--compile_flags=-O2 -g -Wall -fdiagnostics-show-option -Werror  -Wno-unused-function -Wno-char-subscripts -Wno-c++11-extensions -Wno-unnamed-type-template-args -Wno-extra-semi -Wno-unused-private-field -Wno-char-subscripts -Wno-unused-function \"-std=gnu++11\" " --gomadir /b/build/goma "--defines=\"__STDC_LIMIT_MACROS=1\" \"__STDC_FORMAT_MACROS=1\" \"_GNU_SOURCE=1\" \"_POSIX_C_SOURCE=199506\" \"_XOPEN_SOURCE=600\" \"DYNAMIC_ANNOTATIONS_ENABLED=1\" \"DYNAMIC_ANNOTATIONS_PREFIX=NACL_\" \"NACL_BUILD_ARCH=x86\" V8_DEPRECATION_WARNINGS \"CLD_VERSION=2\" \"_FILE_OFFSET_BITS=64\" CHROMIUM_BUILD \"CR_CLANG_REVISION=255169-1\" COMPONENT_BUILD UI_COMPOSITOR_IMAGE_TRANSPORT \"USE_AURA=1\" \"USE_ASH=1\" \"USE_PANGO=1\" \"USE_CAIRO=1\" \"USE_DEFAULT_RENDER_THEME=1\" \"USE_LIBJPEG_TURBO=1\" \"USE_X11=1\" \"IMAGE_LOADER_EXTENSION=1\" \"ENABLE_WEBRTC=1\" \"ENABLE_MEDIA_ROUTER=1\" USE_PROPRIETARY_CODECS ENABLE_PEPPER_CDMS ENABLE_CONFIGURATION_POLICY ENABLE_NOTIFICATIONS \"ENABLE_HIDPI=1\" \"ENABLE_TOPCHROME_MD=1\" USE_UDEV DONT_EMBED_BUILD_METADATA \"DCHECK_ALWAYS_ON=1\" FIELDTRIAL_TESTING_ENABLED \"ENABLE_TASK_MANAGER=1\" \"ENABLE_EXTENSIONS=1\" \"ENABLE_PDF=1\" \"ENABLE_PLUGINS=1\" \"ENABLE_SESSION_SERVICE=1\" \"ENABLE_THEMES=1\" \"ENABLE_AUTOFILL_DIALOG=1\" \"ENABLE_BACKGROUND=1\" \"ENABLE_PRINTING=1\" \"ENABLE_PRINT_PREVIEW=1\" \"ENABLE_SPELLCHECK=1\" \"ENABLE_CAPTIVE_PORTAL_DETECTION=1\" \"ENABLE_APP_LIST=1\" \"ENABLE_SUPERVISED_USERS=1\" \"ENABLE_MDNS=1\" \"ENABLE_SERVICE_DISCOVERY=1\" V8_USE_EXTERNAL_STARTUP_DATA FULL_SAFE_BROWSING SAFE_BROWSING_CSD SAFE_BROWSING_DB_LOCAL EXPAT_RELATIVE_PATH FEATURE_ENABLE_SSL GTEST_RELATIVE_PATH HAVE_OPENSSL_SSL_H NO_MAIN_THREAD_WRAPPING NO_SOUND_SYSTEM WEBRTC_POSIX SRTP_RELATIVE_PATH SSL_USE_OPENSSL USE_WEBRTC_DEV_BRANCH \"timezone=_timezone\" XML_STATIC \"USE_LIBPCI=1\" \"USE_OPENSSL=1\" \"USE_OPENSSL_CERTS=1\"" "--link_flags=-B../../out/Debug/gen/tc_pnacl_newlib/lib  " "--source-list=../../out/gypfiles/third_party/libjingle/pnacl_newlib.libjingle_nacl.source_list.gypcmd"
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:50:23: error: 'CreateConnection' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
  virtual Connection* CreateConnection(const Candidate& address,
                      ^
../webrtc/p2p/base/portinterface.h:71:23: note: overridden virtual function is here
  virtual Connection* CreateConnection(
                      ^
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:53:16: error: 'PrepareAddress' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
  virtual void PrepareAddress();
               ^
../webrtc/p2p/base/portinterface.h:63:16: note: overridden virtual function is here
  virtual void PrepareAddress() = 0;
               ^

(etc)

Original issue's description:
> Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
>
> To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
>
> BUG=4173
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/7307952a5bf63311e5f9a2a90089a06177e42716
> Cr-Commit-Position: refs/heads/master@{#11247}

TBR=pthatcher@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=4173

Review URL: https://codereview.webrtc.org/1586063002

Cr-Commit-Position: refs/heads/master@{#11249}
2016-01-14 12:57:03 +00:00
Stefan Holmer
7307952a5b Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.

BUG=4173
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1577873003 .

Cr-Commit-Position: refs/heads/master@{#11247}
2016-01-14 12:15:56 +00:00
honghaiz
37389b42b4 Don't delete an ICE connection until it has been pruned or timed out on writing in the case where it
hasn't received anything yet.  Deleting an ICE connection before it is pruned or timed out
when it hasn't received anything yet leads to ICE connections being deleted
before they have a chance to send a ping and receive a response.
BUG=

Review URL: https://codereview.webrtc.org/1544003002

Cr-Commit-Position: refs/heads/master@{#11151}
2016-01-05 05:57:42 +00:00
Honghai Zhang
f9945b2d1a Only try to pair protocol matching candidates for creating connections.
If the local port and the remote candidate's protocols do not match,
do not even try to pair them.
This avoids printing out confusing logs like
"Attempt to change a remote candidate..." in p2ptransportchannel
when two remote candidates have the same port number but different
protocols.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1516613002 .

Cr-Commit-Position: refs/heads/master@{#11034}
2015-12-15 20:20:22 +00:00
pbos
7640ffabd7 Initialize type_preference_ in TestPort.
Prevents use of undefined memory for logging during
PortTest.TestLoopbackCal which was recently enabled for all release
builds.

BUG=
R=asapersson@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1480233003

Cr-Commit-Position: refs/heads/master@{#10842}
2015-11-30 17:17:07 +00:00
Honghai Zhang
2cd7afe7e2 Do not delete a connection until it has not received anything for 30 seconds.
BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1422623015 .

Cr-Commit-Position: refs/heads/master@{#10626}
2015-11-12 19:14:38 +00:00
tfarina
5237aaf243 Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
2015-11-11 07:44:39 +00:00
Guo-wei Shieh
9af97f8910 WebRTC should generate default private address even when adapter enumeration is disabled.
Introduce a DefaultAddressProvider such that rtc::Network can't access other part of NetworkManager.

This also removes the hack of generating the loopback address. The dependency has been removed by https://codereview.chromium.org/1417023003/

BUG=webrtc:5061
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1411253008 .

Cr-Commit-Position: refs/heads/master@{#10590}
2015-11-10 22:47:49 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
honghaiz
d0b3143f0e Do not time out a port if its role switched from controlled to controlling. Also fix some comments.
BUG=webrtc:5026

Review URL: https://codereview.webrtc.org/1376983002

Cr-Commit-Position: refs/heads/master@{#10122}
2015-09-30 19:42:25 +00:00
Peter Thatcher
04ac81f2fd Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
BUG=4937
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1345913004 .

Cr-Commit-Position: refs/heads/master@{#10004}
2015-09-21 18:48:37 +00:00
tommi
275a2f16fd Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ )
Reason for revert:
Broke the Windows build:

[226/365] LINK_EMBED cc_perftests.exe
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj.rsp /c ..\..\remoting\protocol\channel_socket_adapter_unittest.cc /Foobj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj /Fdobj\remoting\remoting_unittests.cc.pdb
e:\b\build\slave\win\build\src\remoting\protocol\channel_socket_adapter_unittest.cc(36) : error C3861: 'set_readable': identifier not found
ninja: build stopped: subcommand failed.

Original issue's description:
> Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
> If a connection does not receive for 30 seconds, it will be deleted.
> BUG=
>
> Committed: https://crrev.com/ae16f8547d3b447f62f6660f13688585c6c3de15
> Cr-Commit-Position: refs/heads/master@{#10001}

TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1356103002

Cr-Commit-Position: refs/heads/master@{#10002}
2015-09-21 14:20:43 +00:00
honghaiz
ae16f8547d Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
If a connection does not receive for 30 seconds, it will be deleted.
BUG=

Review URL: https://codereview.webrtc.org/1351673003

Cr-Commit-Position: refs/heads/master@{#10001}
2015-09-21 13:54:19 +00:00
Peter Thatcher
7cbd188c5e Remove GICE (again).
R=guoweis@webrtc.org

Review URL: https://codereview.webrtc.org/1353713002 .

Cr-Commit-Position: refs/heads/master@{#9979}
2015-09-18 01:55:03 +00:00
guoweis
d12140a68e Revert change which removes GICE.
There are still dependencies on this functionality.

TBR=pthatcher@webrtc.org

BUG=526399

Review URL: https://codereview.webrtc.org/1336553003

Cr-Commit-Position: refs/heads/master@{#9920}
2015-09-10 20:32:21 +00:00
Guo-wei Shieh
1eb87c7d94 TCPConnection can never be deteted if they fail to connect.
Since the TCPConnection has never been connected, they are not scheduled for ping hence will never be detected.

Also fix the case when reconnect fails, as it has become READABLE before, it also will not be deleted.

BUG=webrtc:4936
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1307083002 .

Cr-Commit-Position: refs/heads/master@{#9782}
2015-08-25 18:03:02 +00:00
Guo-wei Shieh
b594041ec8 TcpPort Reconnect should inform upper layer to start sending again.
During the reconnection phase, EWOULDBLOCK has been returned to upper layer which stops the sending of video stream.

BUG=webrtc:4930
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1288553010 .

Cr-Commit-Position: refs/heads/master@{#9767}
2015-08-24 18:58:07 +00:00
Peter Thatcher
2159b89fa2 Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 5bdafd44c86ee46bd7e040f19828324583418b33.

Original CL: https://codereview.webrtc.org/1263663002/

R=guoweis@webrtc.org

Review URL: https://codereview.webrtc.org/1303393002 .

Cr-Commit-Position: refs/heads/master@{#9761}
2015-08-22 03:46:18 +00:00
minyuel
5bdafd44c8 Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""
This reverts commit 081f34b564e1a26ffbbe9515eba1fef7c736fdde.

Original code review see
https://codereview.webrtc.org/1291363005

The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137

TBR=pthatcher@webrtc.org,

BUG=

Review URL: https://codereview.webrtc.org/1308753003 .

Cr-Commit-Position: refs/heads/master@{#9756}
2015-08-21 13:52:58 +00:00
Guo-wei Shieh
fe3bc9d5ae Relanding "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
Migrated from https://codereview.webrtc.org/1275703006/ which causes test failures for android. On android, loopback interface was used as local interface to generate candidates. Add a test case to make sure this won't be broken in the future.

Also observed some failures under content_browsertests in chromium.fyi bot but can't repro locally. Might just be temporary test issue.

BUG=webrtc:4517
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1299333003 .

Cr-Commit-Position: refs/heads/master@{#9746}
2015-08-20 15:48:27 +00:00
Peter Thatcher
081f34b564 Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.

R=guoweis@webrtc.org

Review URL: https://codereview.webrtc.org/1291363005 .

Cr-Commit-Position: refs/heads/master@{#9738}
2015-08-20 03:37:59 +00:00
Guo-wei Shieh
3d564c1015 Add instrumentation to track the IceEndpointType.
The IceEndpointType has the format of <local_endpoint>_<remote_endpoint>. It is recorded on the BestConnection when we have the first OnTransportCompleted signaled.

BUG=webrtc:4918
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1277263002 .

Cr-Commit-Position: refs/heads/master@{#9737}
2015-08-19 23:51:22 +00:00
Guo-wei Shieh
370c8848ad Revert "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
This reverts commit 0a2955f227666efd87b2a303a69c083ef801c528.

Revert "In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate."

This reverts commit ba9ab4cd8d2e8fbc068dc36b5e6f6331d7deeccf.

TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1288843003 .

Cr-Commit-Position: refs/heads/master@{#9729}
2015-08-19 00:00:21 +00:00
Guo-wei Shieh
0a2955f227 Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied.
BUG=webrtc:4517
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1275703006 .

Cr-Commit-Position: refs/heads/master@{#9726}
2015-08-18 20:05:29 +00:00
pthatcher
fa301809b6 Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.

TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1274273005

Cr-Commit-Position: refs/heads/master@{#9698}
2015-08-11 11:13:00 +00:00
Peter Thatcher
3449faa553 Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
R=deadbeef@webrtc.org, juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1263663002 .

Cr-Commit-Position: refs/heads/master@{#9692}
2015-08-10 19:22:59 +00:00