Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 3544d3031a90e44779d593495a9d3eb4ee091047
https://github.com/WebKit/WebKit/commit/3544d3031a90e44779d593495a9d3eb4ee091047
Author: Philippe Normand <[email protected]>
Date: 2023-04-19 (Wed, 19 Apr 2023)
Changed paths:
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h
Log Message:
-----------
[GStreamer][WebRTC] RTP payload type numbers are incorrect and prone to
collisions in our SDP offers
https://bugs.webkit.org/show_bug.cgi?id=255614
Reviewed by Xabier Rodriguez-Calvar.
Use static payload types for the legacy audio codecs, as mandated by their
respective RTP payload
RFCs and for dynamic payloads, use unique identifiers, starting at 96. See also
https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::requestPad):
(WebCore::GStreamerMediaEndpoint::pickAvailablePayloadType):
(WebCore::GStreamerMediaEndpoint::createTransceiverBackends):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h:
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::payloadTypeForEncodingName):
(WebCore::capsFromRtpCapabilities):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.h:
Canonical link: https://commits.webkit.org/263135@main
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