Confirmed users
147
edits
(Filled webrtc networking section) |
(Filled webrtc signaling section) |
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===WebRTC:Signaling:=== | ===WebRTC:Signaling:=== | ||
{{Bug|1377299}} Add [ChromeOnly] packet dump hooks to RTCPeerConnection | |||
{{Bug|1392961}} Make VP9 the default decoder/encoder by default | |||
{{Bug|1401540}} InvalidSessionDescriptionError: "Empty BUNDLE group" | |||
{{Bug|1403204}} Code that configures telephone-event assumes that it is always last in the codec array | |||
{{Bug|1405940}} Crash - WebRtc - Null Pointer dereference in sigslot::lock_block | |||
{{Bug|1408523}} permaleak of 288 bytes of thread stuff in dom/media/tests/mochitest/test_peerConnection_basicH264Video.html | |||
{{Bug|1411605}} -Wclass-memaccess: clearing an object of non-trivial type 'struct webrtc::CodecSpecificInfo' | |||
{{Bug|1413709}} add tests to detect improper ice restart from answer changing ufrag/pwd | |||