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Media/WebRTC/ReleaseNotes/41: Difference between revisions
→Signaling:
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===Signaling:=== | ===Signaling:=== | ||
{{Bug|1096795}}: WebRTC SDP does not include required a=rtcp:<port> | {{Bug|1096795}}: WebRTC SDP does not include required a=rtcp:<port> '''- uplifted to 40''' | ||
{{Bug|1155965}}: crash in mozilla::JsepSessionImpl::ValidateLocalDescription(mozilla::Sdp const&) | {{Bug|1155965}}: crash in mozilla::JsepSessionImpl::ValidateLocalDescription(mozilla::Sdp const&) | ||
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{{Bug|1167922}}: Handle broken entries in media.peerconnection.default_iceservers more gracefully | {{Bug|1167922}}: Handle broken entries in media.peerconnection.default_iceservers more gracefully | ||
{{Bug|1142757}}: media/webrtc/signaling/test/mediaconduit_unittests.cpp |AudioSendAndReceive::GenerateAndReadSamples| has several leaks | {{Bug|1142757}}: media/webrtc/signaling/test/mediaconduit_unittests.cpp |AudioSendAndReceive::GenerateAndReadSamples| has several leaks | ||
===WebAudio=== | ===WebAudio=== | ||