| 1172397
|
|
Replaying a HTMLMediaElement streamed over a PeerConnection can crash WebRTC
|
Core
|
WebRTC: Audio/Video
|
apehrson
|
RESO
|
FIXE
|
2016-07-02
|
| 1130061
|
|
Firefox crash in webrtc::videocapturemodule::DeviceInfoDS::CreateCapabilityMap(char const*)
|
Core
|
WebRTC: Audio/Video
|
away
|
RESO
|
FIXE
|
2023-04-18
|
| 1161317
|
|
Incorrect encryption of RTCP Packets when using unidirectional PeerConnections
|
Core
|
WebRTC: Networking
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1167306
|
|
Several webrtc prefs are broken
|
Core
|
WebRTC: Audio/Video
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1182289
|
|
WebrtcGmpVideoEncoder::InitEncode/InitDecode go reentrant on main, opening the door to all kinds of problems
|
Core
|
WebRTC: Audio/Video
|
docfaraday
|
RESO
|
FIXE
|
2017-11-08
|
| 1188590
|
|
jsjni_GetGlobalClassRef goes reentrant on main
|
Core
|
WebRTC: Audio/Video
|
docfaraday
|
RESO
|
FIXE
|
2016-07-02
|
| 1189058
|
|
unresponsive gUM requiring restart on Android after http://webrtc.github.io/samples/src/content/devices/input-output
|
Core
|
WebRTC: Audio/Video
|
gpascutto
|
RESO
|
FIXE
|
2016-07-02
|
| 1166183
|
|
Back out the direct listener removal landed by mistake in bug 1141781
|
Core
|
WebRTC: Audio/Video
|
padenot
|
RESO
|
FIXE
|
2015-06-15
|
| 1151628
|
|
MJPEG getUserMedia sources don't work
|
Core
|
WebRTC: Audio/Video
|
rjesup
|
RESO
|
FIXE
|
2015-06-21
|
| 1152016
|
|
Suppress MJPEG fprintf() warnings in libyuv
|
Core
|
WebRTC: Audio/Video
|
rjesup
|
RESO
|
FIXE
|
2015-04-28
|
| 1157766
|
|
Firefox 37 only supports 16 datachannels
|
Core
|
WebRTC: Networking
|
rjesup
|
RESO
|
FIXE
|
2015-04-30
|
| 1162251
|
|
WebRTC H264 regression in packetization
|
Core
|
WebRTC: Networking
|
rjesup
|
RESO
|
FIXE
|
2015-05-14
|
| 1149494
|
|
video.onloadedmetadata handler doesn't seem to work with MediaStream input since Firefox37
|
Core
|
WebRTC: Audio/Video
|
roc
|
RESO
|
FIXE
|
2015-05-18
|
| 1143694
|
|
WebRTC looks garbled on Lollipop Gonk
|
Core
|
WebRTC
|
sotaro.ikeda.g
|
RESO
|
FIXE
|
2015-10-06
|
| 1131779
|
|
Webrtc stops using relay port after permission error response
|
Core
|
WebRTC: Networking
|
drno
|
RESO
|
FIXE
|
2015-06-10
|
| 1150609
|
|
WebRTC Offer SDP should include tmmbr.
|
Core
|
WebRTC: Signaling
|
ethanhugg
|
RESO
|
FIXE
|
2016-05-04
|
| 1135372
|
|
Intermittent browser_devices_get_user_media_in_frame.js | application crashed [@ mozilla::MediaManager::Observe(nsISupports *,char const *,wchar_t const *)] after "Assertion failure: device, at dom\media\MediaManager.cpp:2058"
|
Core
|
WebRTC
|
nobody
|
RESO
|
FIXE
|
2015-06-11
|
| 1154482
|
|
about:webrtc page occasionally throws a TypeError for an undefined reference
|
Core
|
WebRTC
|
paulrkerr
|
RESO
|
FIXE
|
2015-04-20
|
| 1106958
|
|
[WebRTC] Use android.media.MediaCodec for decoding in WebRTC stack
|
Core
|
WebRTC: Audio/Video
|
qiang.lu
|
RESO
|
FIXE
|
2016-04-11
|
| 1155089
|
|
RTPSender.replaceTrack()ing a gUM audio track with a WebAudio track yields silence
|
Core
|
WebRTC: Audio/Video
|
apehrson
|
RESO
|
FIXE
|
2015-06-10
|
| 1122900
|
|
'alignas' and '__declspec(align)' are not allowed on function declarations
|
Core
|
WebRTC: Audio/Video
|
brian
|
RESO
|
FIXE
|
2015-04-14
|
| 1157995
|
|
Leak with expando on navigator.mediaDevices
|
Core
|
WebRTC
|
continuation
|
RESO
|
FIXE
|
2015-05-24
|
| 1153378
|
|
dom/media/AudioSegment.h:210:18 [-Wunused-variable] unused variable 'in'
|
Core
|
WebRTC
|
cpeterson
|
RESO
|
FIXE
|
2015-04-13
|
| 1127727
|
|
[e10s] Detached tab with shared video is displayed with a large throbber after re-attachment
|
Core
|
WebRTC: Audio/Video
|
davidp99
|
RESO
|
FIXE
|
2015-07-20
|
| 1096795
|
|
WebRTC SDP does not include required a=rtcp:<port>
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1146529
|
|
preferred_codec ignored on answerer side
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1148649
|
|
Reenable video multistream mochitests on debug e10s
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1149298
|
|
null candidate never fires on pc.onicecandidate (REGRESSION)
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1149838
|
|
We should not suppress negotiationneeded before the first offer/answer exchange
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-08-30
|
| 1150966
|
|
Stats methods on NrIceMediaStream do not check whether |stream_| is null
|
Core
|
WebRTC: Networking
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1151139
|
|
Racy call to PeerConnectionMedia::num_ice_media_streams from PeerConnectionImpl::BuildStatsQuery_m
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2016-07-02
|
| 1152093
|
|
WebRTC setRemoteDescription errors out if OPUS is codec in SDP
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1152137
|
|
Numerous UAF bugs in nr_stun_message_add_*_attribute
|
Core
|
WebRTC: Networking
|
docfaraday
|
RESO
|
FIXE
|
2017-05-09
|
| 1155965
|
|
crash in mozilla::JsepSessionImpl::ValidateLocalDescription(mozilla::Sdp const&)
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-07-09
|
| 1160280
|
|
JsepSessionImpl should add ssrc attributes for recvonly m-sections
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1161136
|
|
JsepSessionImpl does not remove mid from bundle group when rejecting an m-section in some cases
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2015-06-10
|
| 1161719
|
|
Possible OOB read in |MediaPipelineFilter::FilterSenderReport|
|
Core
|
WebRTC: Signaling
|
docfaraday
|
RESO
|
FIXE
|
2018-07-06
|
| 1148572
|
|
Re-enabling a H264 m-line results in UAF crash
|
Core
|
WebRTC: Audio/Video
|
drno
|
RESO
|
FIXE
|
2016-07-02
|
| 1161433
|
|
Empty gUM constraints throw wrong error
|
Core
|
WebRTC
|
drno
|
RESO
|
FIXE
|
2015-06-10
|
| 1154389
|
|
Stop leaking DeviceSuccessCallbackRunnable objects, and release its mOnSuccess and mOnFailure on the main thread if needed
|
Core
|
WebRTC
|
ehsan.akhgari
|
RESO
|
FIXE
|
2015-04-15
|
| 1149240
|
|
Unit test for "no digest" condition in mtransport
|
Core
|
WebRTC: Networking
|
ekr
|
RESO
|
FIXE
|
2015-03-31
|
| 1151080
|
|
Rewrite NR_async_schedule to not depend on timers
|
Core
|
WebRTC: Networking
|
ekr
|
RESO
|
FIXE
|
2016-07-02
|
| 1158627
|
|
Crash in VCMJitterBuffer::GetFrame on Windows
|
Core
|
WebRTC: Audio/Video
|
ethanhugg
|
RESO
|
FIXE
|
2015-04-28
|
| 1163239
|
|
GMP Crash at GMPParent::ChildTerminated() on OSX on FF38 on shutdown after using openh264.
|
Core
|
WebRTC: Audio/Video
|
ethanhugg
|
RESO
|
FIXE
|
2015-07-09
|
| 1164061
|
|
WebRTC Move TMMBR use behind pref.
|
Core
|
WebRTC: Audio/Video
|
ethanhugg
|
RESO
|
FIXE
|
2015-09-22
|
| 1159840
|
|
Remove libvpx alignment patch
|
Core
|
WebRTC
|
giles
|
RESO
|
FIXE
|
2015-05-06
|
| 1155140
|
|
documentation for test helper functions mention removed parameters
|
Core
|
WebRTC
|
gordon.klaus
|
RESO
|
FIXE
|
2015-04-17
|
| 1155648
|
|
documentation for DOMMediaStream::OnTracksAvailable is stale
|
Core
|
WebRTC
|
gordon.klaus
|
RESO
|
FIXE
|
2015-04-27
|
| 1152443
|
|
enumerateDevices is not persisting IDs after a restart
|
Core
|
WebRTC: Audio/Video
|
jib
|
RESO
|
FIXE
|
2015-04-09
|
| 1153056
|
|
about:webrtc blanks whenever allocated PeerConnections goes to zero (REGRESSION)
|
Core
|
WebRTC: Signaling
|
jib
|
RESO
|
FIXE
|
2015-05-10
|
| 1162720
|
|
enumerateDevices crashes on Android (webcamtoy.com crashes)
|
Core
|
WebRTC: Audio/Video
|
jib
|
RESO
|
FIXE
|
2015-06-19
|
| 1146637
|
|
Firefox OS (flame) ignores constraints, chooses very low video resolution
|
Core
|
WebRTC: Audio/Video
|
jryans
|
RESO
|
FIXE
|
2015-04-20
|
| 1156621
|
|
USE_LIBS contains "/nsprpub/lib/libc/src/plc4", but there is no "plc4" LIBRARY_NAME in nsprpub/lib/libc/src --with-system-nspr
|
Core
|
WebRTC
|
karlt
|
RESO
|
FIXE
|
2015-04-22
|
| 1160745
|
|
Fix webrtc tests building on OpenBSD (missing -lsndio)
|
Core
|
WebRTC
|
landry
|
RESO
|
FIXE
|
2015-05-26
|
| 996238
|
|
ALPN identifiers for WebRTC
|
Core
|
WebRTC: Networking
|
mt
|
RESO
|
FIXE
|
2015-06-10
|
| 1152538
|
|
Enable WebRTC identity by default
|
Core
|
WebRTC
|
mt
|
RESO
|
FIXE
|
2016-03-18
|
| 1153314
|
|
XHR from IdP sandbox fails
|
Core
|
WebRTC: Signaling
|
mt
|
RESO
|
FIXE
|
2015-06-10
|
| 1158931
|
|
Static assertion - WebrtcAndroidMediaCodec should not have a public destructor
|
Core
|
WebRTC: Audio/Video
|
nathan
|
RESO
|
FIXE
|
2015-04-28
|
| 1100502
|
|
about:webrtc Start Debug Log/etc don't work on Nightly (e10s)
|
Core
|
WebRTC
|
paulrkerr
|
RESO
|
FIXE
|
2017-10-14
|
| 1159320
|
|
--disable-webrtc builds now error out with "dom/ipc/PContent.ipdl:53: error: can't locate include file `PWebrtcGlobal.ipdl'"
|
Core
|
WebRTC
|
paulrkerr
|
RESO
|
FIXE
|
2017-10-14
|
| 1161619
|
|
|RunStatsQuery| leaks |queries|
|
Core
|
WebRTC: Signaling
|
paulrkerr
|
RESO
|
FIXE
|
2018-07-06
|
| 1093934
|
|
Create a XPCOM library that can be used to support standalone WebRTC
|
Core
|
WebRTC
|
rbarker
|
RESO
|
FIXE
|
2016-01-06
|
| 1097804
|
|
Create a library containing nsISocketTransportService and nsIDNS that can be used to support standalone WebRTC
|
Core
|
WebRTC
|
rbarker
|
RESO
|
FIXE
|
2017-02-15
|
| 1101651
|
|
Enable WebRTC unit tests to be built using standalone WebRTC library
|
Core
|
WebRTC
|
rbarker
|
RESO
|
FIXE
|
2015-06-25
|
| 1132318
|
|
refine resolution/framerate selection
|
Core
|
WebRTC
|
rjesup
|
RESO
|
FIXE
|
2015-06-09
|
| 1159300
|
|
GMP OpenH264 fails to decode on reload on Windows
|
Core
|
WebRTC: Audio/Video
|
rjesup
|
RESO
|
FIXE
|
2015-05-13
|
| 1161079
|
|
Logging of encoder stats in WebRTC broken by multistream landing
|
Core
|
WebRTC
|
rjesup
|
RESO
|
FIXE
|
2019-10-25
|
| 1163852
|
|
Intermittent test_peerConnection_addSecondVideoStream.html | application terminated with exit code -5
|
Core
|
WebRTC
|
rjesup
|
RESO
|
FIXE
|
2016-07-02
|
| 1150271
|
|
WebRTC session crashes in mozilla::MediaEngineGonkVideoSource::StartImpl()
|
Core
|
WebRTC: Audio/Video
|
sotaro.ikeda.g
|
RESO
|
FIXE
|
2015-04-17
|
| 1153050
|
|
Add ICameraControl check to MediaEngineGonkVideoSource::Start()
|
Core
|
WebRTC: Audio/Video
|
sotaro.ikeda.g
|
RESO
|
FIXE
|
2015-04-13
|
| 1158868
|
|
PeerConnection should not depend on navigator.onLine
|
Core
|
WebRTC
|
valentin.gosu
|
RESO
|
FIXE
|
2015-04-29
|
| 1181265
|
|
crash in webrtc::videocapturemodule::DeviceInfoDS::CreateCapabilityMap(char const*)
|
Core
|
WebRTC: Audio/Video
|
rjesup
|
VERI
|
FIXE
|
2015-08-24
|
| 1157701
|
|
test_peerConnection_replaceTrack.html is going to permafail when Gecko 40 merges to Aurora
|
Core
|
WebRTC: Audio/Video
|
apehrson
|
VERI
|
FIXE
|
2015-06-10
|
| 1137614
|
|
crash in vp8_diamond_search_sadx4
|
Core
|
WebRTC
|
away
|
VERI
|
FIXE
|
2015-06-23
|
| 1150539
|
|
getUserMedia activating camera light, but not receiving video -- audio working OK.
|
Core
|
WebRTC: Audio/Video
|
jib
|
VERI
|
FIXE
|
2015-07-06
|
| 1162412
|
|
facingMode regression in 38
|
Core
|
WebRTC: Audio/Video
|
jib
|
VERI
|
FIXE
|
2015-06-23
|
| 1159659
|
|
Allow browser sharing gUM requests on Windows XP and OSX 10.6
|
Core
|
WebRTC
|
mreavy
|
VERI
|
FIXE
|
2015-05-15
|