From: Joshua Colp <jcolp@digium.com>
Date: Mon, 7 Jan 2019 14:06:37 +0000 (+0000)
Subject: res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.
X-Git-Tag: 16.2.0-rc1~25^2
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=c6271155fb01700eacddd20651ba9765fabce194;p=thirdparty%2Fasterisk.git

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
---

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");