[webkit-changes] [WebKit/WebKit] 3544d3: [GStreamer][WebRTC] RTP payload type numbers are i...

[Philippe Normand]

  Branch: refs/heads/main
  Home:   https://github.com/WebKit/WebKit
  Commit: 3544d3031a90e44779d593495a9d3eb4ee091047
      https://github.com/WebKit/WebKit/commit/3544d3031a90e44779d593495a9d3eb4ee091047
  Author: Philippe Normand <philn at igalia.com>
  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