| 998546
|
WebRTC
|
P1
|
15
|
docfaraday
|
RESO
|
2016-05-13
|
ontrack and onaddstream fire too late
|
| 1019579
|
WebRTC: Audio/Video
|
P1
|
15
|
docfaraday
|
RESO
|
2016-05-16
|
PeerConnection MUST include all remote tracks after SetRemoteDescriptionSuccessCallback
|
| 1213773
|
WebRTC
|
P1
|
17
|
docfaraday
|
RESO
|
2016-12-07
|
properly handle a=inactive in the remote SDP during renegotiation
|
| 1231975
|
WebRTC: Networking
|
P1
|
15
|
docfaraday
|
RESO
|
2016-05-02
|
Mochitests for NAT scenarios
|
| 1231981
|
WebRTC: Networking
|
P1
|
15
|
docfaraday
|
RESO
|
2017-01-30
|
Test TURN server for mochitest in CI
|
| 1273136
|
WebRTC: Audio/Video
|
P1
|
15
|
docfaraday
|
RESO
|
2016-06-06
|
PeerConnection shouldn't expose a received MediaStreamTrack without guaranteeing that the underlying track goes live
|
| 1251821
|
WebRTC: Networking
|
P1
|
12
|
drno
|
RESO
|
2016-04-27
|
[WebRtc] Packet loss when sending/receiving RTP stream (1080P) in windows 7
|
| 1268291
|
WebRTC: Networking
|
P1
|
10
|
drno
|
RESO
|
2016-05-06
|
ICE Consent request are missing priority and controlled
|
| 1269486
|
WebRTC: Networking
|
P1
|
10
|
drno
|
RESO
|
2016-05-05
|
ICE role switch when answerer starts renegotiation
|
| 1155435
|
WebRTC: Audio/Video
|
P1
|
15
|
mfroman
|
RESO
|
2016-11-22
|
WebRTC - investigate enabling REMB
|
| 1264470
|
WebRTC
|
P1
|
17
|
mt
|
RESO
|
2016-05-02
|
a=identity attribute is truncated, duplicated
|
| 1268428
|
WebRTC: Audio/Video
|
P1
|
17
|
padenot
|
RESO
|
2016-05-10
|
Disable audio down/up sampling when all the constraints have been passed in gUM
|
| 1207431
|
WebRTC: Audio/Video
|
P1
|
10
|
rjesup
|
RESO
|
2016-05-29
|
Intermittent leakcheck | default process: 600 bytes leaked (CondVar, Mutex, nsRunnable, nsTArray_base, nsThread, ...)
|
| 1213050
|
WebRTC
|
P1
|
15
|
rjesup
|
RESO
|
2016-05-27
|
Intermittent test_zmedia_cleanup.html | application crashed [@ mozilla::(anonymous namespace)::RunWatchdog(void*)]
|
| 1240209
|
WebRTC: Networking
|
P1
|
15
|
rjesup
|
RESO
|
2016-06-14
|
Increased latency with WebRTC data channels
|
| 1243857
|
WebRTC: Audio/Video
|
P1
|
7
|
rjesup
|
RESO
|
2016-10-06
|
Enable full-duplex cubeb backends for Desktop
|
| 1269930
|
WebRTC: Audio/Video
|
P1
|
10
|
rjesup
|
RESO
|
2017-03-07
|
Crash on windows when logging AEC data from about:webrtc
|
| 1273206
|
WebRTC: Audio/Video
|
P1
|
17
|
rjesup
|
RESO
|
2016-09-16
|
enumDev/gUM starts RTP thread in 5ms loop
|
| 1275703
|
WebRTC: Audio/Video
|
P1
|
10
|
rjesup
|
RESO
|
2016-05-26
|
AEC is off in second and later getUserMedia() calls
|
| 1276156
|
WebRTC
|
P1
|
15
|
rjesup
|
RESO
|
2016-06-01
|
deadlock shutting down vie_encoder in statistics
|
| 1206708
|
WebRTC: Audio/Video
|
P2
|
22
|
apehrson
|
RESO
|
2016-05-13
|
WebRTC video is doing copying video frames more than it needs to
|
| 1271669
|
WebRTC
|
P2
|
25
|
apehrson
|
RESO
|
2016-08-09
|
Allow any MediaStream to be passed to RTCPeerConnection.addTrack
|
| 1180725
|
WebRTC: Audio/Video
|
P2
|
27
|
bonchiang
|
RESO
|
2016-12-15
|
Use AVFoundation for camera capture on OSX
|
| 1276383
|
WebRTC
|
P2
|
23
|
continuation
|
RESO
|
2016-06-09
|
Add nsIAsyncShutdown.xpcomWillShutdown and use it in WebRTC
|
| 929977
|
WebRTC: Networking
|
P2
|
26
|
drno
|
RESO
|
2016-04-27
|
ICE consent freshness (RFC 7675) not implemented
|
| 1271862
|
WebRTC: Signaling
|
P2
|
29
|
drno
|
RESO
|
2016-05-12
|
Fail to increase SDP session version in renegotiated answer
|
| 1276368
|
WebRTC
|
P2
|
25
|
drno
|
RESO
|
2016-06-06
|
Implement receiving 'b=TIAS' on media description to restrict sending bandwidth
|
| 802326
|
WebRTC
|
P2
|
21
|
jib
|
RESO
|
2017-01-23
|
If video and audio is requested in gUM, but one of them fails, we should align with the spec
|
| 1257950
|
WebRTC: Audio/Video
|
P2
|
25
|
jib
|
RESO
|
2016-06-26
|
getUserMedia spec switched from SecurityError (for permission denied) to NotAllowedError.
|
| 1269268
|
WebRTC
|
P2
|
25
|
khuey
|
RESO
|
2016-05-05
|
Intermittent leakcheck | default process: 212 bytes leaked (CondVar, Mutex, Runnable, nsTArray_base, nsThread)
|
| 1271429
|
WebRTC: Signaling
|
P2
|
25
|
rjesup
|
RESO
|
2018-04-13
|
sdp_unittests doesn't Shutdown() the pseudo-main thread before finishing
|
| 1268449
|
WebRTC: Networking
|
P3
|
35
|
bpostelnicu
|
RESO
|
2016-05-26
|
[Static Analysis][Resource leak] In function Resolve from ice_unittest.cpp
|
| 1273965
|
WebRTC
|
P3
|
35
|
dholbert
|
RESO
|
2016-05-19
|
MediaPipelineFactory.cpp:1015:10: warning: code will never be executed [-Wunreachable-code]
|
| 1275217
|
WebRTC: Audio/Video
|
P3
|
35
|
drno
|
RESO
|
2016-11-30
|
Remove QTKit dependency
|
| 1266468
|
WebRTC: Networking
|
--
|
|
docfaraday
|
RESO
|
2016-04-26
|
Assertion in nr_ice_media_stream_start_checks caused by incoming STUN request after ICE failure
|
| 1269165
|
WebRTC: Audio/Video
|
--
|
|
jbeich
|
RESO
|
2016-05-20
|
getUserMedia fails to enumerate ALSA plugins
|
| 1271041
|
WebRTC
|
--
|
|
nobody
|
RESO
|
2016-05-10
|
Switch NetBSD to pthread_condattr_setclock(,CLOCK_MONOTONIC) in WebRTC
|