dont allocate a payload type for rtp data channels when using sctp
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}
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 6d8a9a4..4fd3efa 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -1512,8 +1512,11 @@
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 @@
// 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:
