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dont allocate a payload type for rtp data channels when using sctp

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BUG=webrtc:12194,webrtc:6625

Change-Id: Ifc8f0b197a536626c16ba5c3ebccbf242008c857
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193861
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32651}
This commit is contained in:
Philipp Hancke
2020-11-17 20:42:54 +01:00
committed by Commit Bot
parent c45f68a026
commit 766a32c28d
1 changed files with 9 additions and 3 deletions
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12
pc/media_session.cc
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@ -1512,8 +1512,11 @@ std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer(
AudioCodecs offer_audio_codecs;
VideoCodecs offer_video_codecs;
RtpDataCodecs offer_rtp_data_codecs;
GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
&offer_video_codecs, &offer_rtp_data_codecs);
GetCodecsForOffer(
current_active_contents, &offer_audio_codecs, &offer_video_codecs,
session_options.data_channel_type == DataChannelType::DCT_SCTP
? nullptr
: &offer_rtp_data_codecs);
if (!session_options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(&offer_audio_codecs);
@ -1930,7 +1933,10 @@ void MediaSessionDescriptionFactory::GetCodecsForOffer(
// Add our codecs that are not in the current description.
MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes);
MergeCodecs<VideoCodec>(all_video_codecs_, video_codecs, &used_pltypes);
MergeCodecs<DataCodec>(rtp_data_codecs_, rtp_data_codecs, &used_pltypes);
// Only allocate a payload type for rtp datachannels when using rtp data
// channels.
if (rtp_data_codecs)
MergeCodecs<DataCodec>(rtp_data_codecs_, rtp_data_codecs, &used_pltypes);
}
// Getting codecs for an answer involves these steps: