sysmocom
/
asterisk
11
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Merge "res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled." into 16

This commit is contained in:
Joshua C. Colp 2019-01-14 08:04:10 -06:00 committed by Gerrit Code Review
parent 32934180f9 c6271155fb
commit 11d63c9e4f
1 changed files with 3 additions and 1 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
res/res_pjsip_sdp_rtp.c
View File

@ -272,7 +272,9 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_RECV, session->endpoint->media.webrtc);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_SEND, session->endpoint->media.webrtc);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_REMB, session->endpoint->media.webrtc);
enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
if (session->endpoint->media.webrtc) {
enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
}
if (session->endpoint->media.tos_video || session->endpoint->media.cos_video) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
session->endpoint->media.cos_video, "SIP RTP Video");