From 18e206381ab2d87da0278392883c59a2dc7122f1 Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Mon, 7 Jan 2019 14:06:37 +0000 Subject: [PATCH] res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled. For video streams it was possible for the abs-send-time information to be placed into RTP streams even if not negotiated. Depending on the endpoint in use this could cause video to not flow. We now only enable abs-send-time for negotiation if WebRTC is enabled. ASTERISK-28230 Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c --- res/res_pjsip_sdp_rtp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 288961554b..c18627ed3d 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -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");