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Steve Anton 07563732f6 [Unified Plan] Avoid offering two senders with the same ID 
This can happen with the following sequence of API calls:
1) AddTrack(track) + offer/answer
2) RemoveTrack(track's sender) + offer/answer
3) AddTrack(same track)

Since the first transceiver had already been used to send, it will
not get re-used by the second call to AddTrack. Another RtpSender
will be created with its ID = the track ID. But the code hits a
DCHECK when CreateOffer is later called since both m= sections will
offer the same track ID component of the MSID.

The fix implemented here is to randomly generate a sender ID if
there is already an RtpSender with the track's ID.

Bug: webrtc:8734
Change-Id: Ic2dda23d66e364e77ff7505e1c37e53105a17dae
Reviewed-on: https://webrtc-review.googlesource.com/84249
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23748}
2018-06-26 19:06:17 +00:00
..
test
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
audiotrack.cc
Remove "using rtc::scoped_ptr" from audiotrack.cc.
2017-11-28 21:29:18 +00:00
audiotrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
BUILD.gn
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
bundlefilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.h
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport.""
2018-03-29 02:45:17 +00:00
channel_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
channel.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
channel.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
channelmanager_unittest.cc
Delete unused include of fakevideocapturer.h.
2018-05-17 07:23:01 +00:00
channelmanager.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
channelmanager.h
Use new TransportController implementation in PeerConnection.
2018-03-30 18:41:19 +00:00
createpeerconnectionfactory.cc
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
datachannel_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
datachannel.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
datachannel.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
DEPS
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
dtlssrtptransport_unittest.cc
Add a flag to actively reset the SRTP parameters
2018-06-12 20:32:00 +00:00
dtlssrtptransport.cc
Add a flag to actively reset the SRTP parameters
2018-06-12 20:32:00 +00:00
dtlssrtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
dtmfsender_unittest.cc
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
dtmfsender.cc
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
dtmfsender.h
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
externalhmac.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
externalhmac.h
[CFI] Fix cfi-icall failures
2017-12-12 09:27:07 +00:00
iceserverparsing_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
iceserverparsing.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
iceserverparsing.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
jsepicecandidate.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
jsepsessiondescription_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
jsepsessiondescription.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransportcontroller_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransportcontroller.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
jseptransportcontroller.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
localaudiosource_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
localaudiosource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
localaudiosource.h
Delete deprecated CreateAudioSource method, with constraints.
2018-06-04 08:19:30 +00:00
mediaconstraintsinterface_unittest.cc
Rename a few MediaConfig::Video flags for consistency.
2018-01-22 17:32:58 +00:00
mediasession_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediasession.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
mediasession.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamobserver.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamobserver.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamtrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
OWNERS
Clean up the pc/OWNERS file.
2018-02-03 19:28:58 +00:00
peerconnection_bundle_unittest.cc
Built in video codec factories.
2018-05-03 11:49:42 +00:00
peerconnection_crypto_unittest.cc
Built in video codec factories.
2018-05-03 11:49:42 +00:00
peerconnection_datachannel_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnection_histogram_unittest.cc
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnection_ice_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnection_integrationtest.cc
Fixing flakiness in PeerConnectionIntegrationTest.
2018-06-19 22:34:53 +00:00
peerconnection_jsep_unittest.cc
[Unified Plan] Do not initialize recvonly transceivers with any send streams
2018-06-26 17:41:56 +00:00
peerconnection_media_unittest.cc
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
peerconnection_rtp_unittest.cc
[Unified Plan] Avoid offering two senders with the same ID
2018-06-26 19:06:17 +00:00
peerconnection_signaling_unittest.cc
Built in video codec factories.
2018-05-03 11:49:42 +00:00
peerconnection.cc
[Unified Plan] Avoid offering two senders with the same ID
2018-06-26 19:06:17 +00:00
peerconnection.h
Refactor RtpSender to take the sender ID as a constructor argument
2018-06-25 21:01:02 +00:00
peerconnectionendtoend_unittest.cc
Re-enable PeerConnectionEndToEndTest.Call on TSan.
2018-06-25 15:36:02 +00:00
peerconnectionfactory_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
peerconnectionfactory.cc
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnectionfactory.h
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
2018-06-25 10:25:38 +00:00
peerconnectioninterface_unittest.cc
[Unified Plan] Avoid offering two senders with the same ID
2018-06-26 19:06:17 +00:00
peerconnectioninternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnectionwrapper.cc
Convert PeerConnectionWrapper from FakeVideoCapturer to FakeVideoTrackSource.
2018-05-30 09:17:17 +00:00
peerconnectionwrapper.h
Convert PeerConnectionWrapper from FakeVideoCapturer to FakeVideoTrackSource.
2018-05-30 09:17:17 +00:00
proxy_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remoteaudiosource.cc
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
remoteaudiosource.h
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
rtcpmuxfilter_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcpmuxfilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcpmuxfilter.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
rtcstats_integrationtest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcstatscollector_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtcstatscollector.cc
Remove stringstreams from pc/
2018-06-11 15:20:59 +00:00
rtcstatscollector.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtcstatstraversal_unittest.cc
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtcstatstraversal.cc
RTCStatsCollector::GetStatsReport() with optional selector argument.
2018-03-19 15:32:16 +00:00
rtcstatstraversal.h
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtpmediautils_unittest.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.h
Reland "Reland "Remove our stream << overloads from non-test build targets.""
2018-05-03 10:41:41 +00:00
rtpreceiver.cc
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
rtpreceiver.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtpsender.cc
Refactor RtpSender to take the sender ID as a constructor argument
2018-06-25 21:01:02 +00:00
rtpsender.h
Refactor RtpSender to take the sender ID as a constructor argument
2018-06-25 21:01:02 +00:00
rtpsenderreceiver_unittest.cc
Refactor RtpSender to take the sender ID as a constructor argument
2018-06-25 21:01:02 +00:00
rtptransceiver.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransceiver.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransportinternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransportinternaladapter.h
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
rtptransporttestutil.h
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
sctputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sdputils.cc
Replace string type with SdpType enum
2017-12-07 19:08:47 +00:00
sdputils.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
sessiondescription.cc
Use the SDP ContentInfo helpers to avoid downcasting
2017-12-22 00:17:53 +00:00
sessiondescription.h
Add MSID signaling compatibility for Unified Plan endpoints
2018-02-01 21:17:41 +00:00
srtpfilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
srtpfilter.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtpfilter.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtpsession_unittest.cc
Report SRTP error codes to UMA
2018-03-20 18:37:49 +00:00
srtpsession.cc
Use RTC_HISTOGRAM_ENUMERATION to report SRTP/SRTCP unprotect error.
2018-06-14 18:35:11 +00:00
srtpsession.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
srtptestutil.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
srtptransport_unittest.cc
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
srtptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
statscollector_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
statscollector.cc
Return SSRC stats with the old stats API when SSRCs are unsignaled.
2018-06-19 17:28:25 +00:00
statscollector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamcollection.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
trackmediainfomap_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
transportstats.cc
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
transportstats.h
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
videocapturertracksource_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videotrack_unittest.cc
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
videotrack.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
videotrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videotracksource.cc
Delete deprecated VideoTrackSource constructor.
2018-06-11 19:15:10 +00:00
videotracksource.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
webrtcsdp_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
webrtcsdp.cc
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
webrtcsdp.h
Add MSID signaling compatibility for Unified Plan endpoints
2018-02-01 21:17:41 +00:00
webrtcsessiondescriptionfactory.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
webrtcsessiondescriptionfactory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00