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

@@ -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");