In some cases, the DTLS ClientHello may arrive before the server's
transport is writable (before it receives a STUN ping response), or
even before it receives a remote fingerprint. If this packet is
discarded, it may take a second for a it to be sent again.
So, this CL caches it instead of dropping it, and feeds it into
the SSL library once the handshake has been started.
BUG=webrtc:5789
Review-Url: https://codereview.webrtc.org/1912323002
Cr-Commit-Position: refs/heads/master@{#12634}
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/1937693002
Cr-Commit-Position: refs/heads/master@{#12581}
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}
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}
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
This propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1920043002
Cr-Commit-Position: refs/heads/master@{#12501}
If a STUN ping arrives before the remote description does, a prflx
candidate will be created with an unknown generation.
Once the remote description does arrive, the candidate's generation
should be set so it can be sorted properly, and replaced by a non-prflx
candidate once the candidate is signaled.
BUG=webrtc:5752
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1868353004 .
Cr-Commit-Position: refs/heads/master@{#12433}
Instead of using a raw pointer output parameter. This affects
SSLStreamAdapter::GetPeerCertificate
Transport::GetRemoteSSLCertificate
TransportChannel::GetRemoteSSLCertificate
TransportController::GetRemoteSSLCertificate
WebRtcSession::GetRemoteSSLCertificate
This is a good idea in general, 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/1802013002
Cr-Commit-Position: refs/heads/master@{#12262}
so that the call knows which packet ids were sent on the previous candidate pair.
Note that packet_id is actually 16bits, so we can use -1 for values that are not set.
Also moved the tests for candidate pair changes to TestSelectConnectionBeforeNomination.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1842093002 .
Cr-Commit-Position: refs/heads/master@{#12184}
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}
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}
No operator== that accepts one unique_ptr<T> and one T*. No implicit
conversion to bool. No rtc_make_scoped_ptr function.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1803833002
Cr-Commit-Position: refs/heads/master@{#12048}
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}
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}
and signaling the remote side to remove its remote candidate by setting the candidate priority to 0.
BUG=
Review URL: https://codereview.webrtc.org/1648813004
Cr-Commit-Position: refs/heads/master@{#11958}
The old code insists on exact cipher suite matches with hardwired expectations. It does this matching parameterized with key type (RSA vs ECDSA) and TLS version (DTLS vs TLS and version 1.0 vs 1.2).
This CL changes things to check against a white-list of cipher suites, with the check parameterized with key type (again RSA vs ECDSA). Then separately checks TLS version since the old implicit check of TLS version by means of resulting cipher suite was too blunt.
Using a white list for cipher suites isn't perfect, but it is safe and requires minimal maintenance. It allows compatibility with not just one exact version of underlying crypto lib, but any version with reasonable defaults.
The CL also re-enables critical tests which had to be disabled recently to allow a boringssl roll.
BUG=webrtc:5634
Review URL: https://codereview.webrtc.org/1774583002
Cr-Commit-Position: refs/heads/master@{#11951}
Also change the type of "time interval" to int from uint32.
Fixed a few TODO therein. I think we should have the following convention:
1. All time delay/intervals should have type int although the time instant should have time uint32_t.
2. "interval" is preferred to "delay" if the delay will be repeated (like rescheduling).
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1762863002 .
Cr-Commit-Position: refs/heads/master@{#11888}
This is useful to keep the NAT binding alive on backup connections.
BUG=
Review URL: https://codereview.webrtc.org/1737493004
Cr-Commit-Position: refs/heads/master@{#11862}
This feature is off by default and can be turned on by setting IceConfig. When turned on, we'll choose a Turn/Turn (UDP takes higher priroity) over the other types of connections while no any connection is writable. However, when there is best connection or there is pending triggered check, those will take higher priority.
BUG=webrtc:4591
Review URL: https://codereview.webrtc.org/1577233006
Cr-Commit-Position: refs/heads/master@{#11850}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.
To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}
This CL will require coordinating landing a roll in Chromium.
BUG=webrtc:4256
NOTRY=True
Review URL: https://codereview.webrtc.org/1750593002
Cr-Commit-Position: refs/heads/master@{#11842}
When remote side closes, opensslstreamadapter could return SR_EOS which will not trigger upper layer to clean up what's left in the StreamInterfaceChannel. The result of this is when there are more packets coming in, the Write on the StreamInterfaceChannel will overflow the buffer.
The fix here is that when receiving the remote side close signal, we also close the underneath StreamInterfaceChannel which will clean up the queue to prevent overflow.
BUG=574524
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1566023002
Cr-Commit-Position: refs/heads/master@{#11751}
RFC 5245 allows an ICE restart to occur on only one media section.
However, before this CL, if an endpoint attempted to do this, we would
change our local ICE ufrag/pwd in every media section.
Also did some refactoring, turning the transport options from
mediasesion.h into a map.
Review URL: https://codereview.webrtc.org/1671173002
Cr-Commit-Position: refs/heads/master@{#11728}
This appears to be dead code because GetTransport() is not used by WebRTC. It also adds dead code to DtlsTransportChannelWrapper and P2PTransportChannel.
BUG=
Review URL: https://codereview.webrtc.org/1691673002
Cr-Commit-Position: refs/heads/master@{#11662}
For now, the network cost is purely based on the network type (cellular has cost 0xFFFF and everything else has cost 0).
Add cost to the candidate signaling and the stun request signaling (which is needed for peer reflexive candidates).
BUG=webrtc:4325
Review URL: https://codereview.webrtc.org/1668073002
Cr-Commit-Position: refs/heads/master@{#11642}
For example, when the TURN port has an ALLOCATE_MISMATCH error.
BUG=webrtc:5432
Review URL: https://codereview.webrtc.org/1595613004
Cr-Commit-Position: refs/heads/master@{#11453}
If it still handle packets, when a ping arrives, it will pass the packet to p2ptransportchannel, eventually causing an ASSERT error there (when p2ptransportchannel tries to create a connection from the ping request from unknown address).
BUG=
Review URL: https://codereview.webrtc.org/1649493006
Cr-Commit-Position: refs/heads/master@{#11430}
It's never used anywhere, so it only causes confusion between
itself and SessionDescriptionInterface::candidates.
Review URL: https://codereview.webrtc.org/1642733002
Cr-Commit-Position: refs/heads/master@{#11420}
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.orgTBR=pthatcher@webtrc.org
BUG=4173
Review URL: https://codereview.webrtc.org/1589563003 .
Cr-Commit-Position: refs/heads/master@{#11251}
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}
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.
BUG=webrtc:4525
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516993002 .
Cr-Commit-Position: refs/heads/master@{#11192}
