Only bitmaskify the RTP payload structure for video if an RTP structure exists...

Only bitmaskify the RTP payload structure for video if an RTP structure exists for it... otherwise the default values will cause codec combination madness


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -4845,7 +4845,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	/* Now gather all of the codecs that we are asked for: */
 	ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
-	ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
+	if (p->vrtp)
+		ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
 	newjointcapability = p->capability & (peercapability | vpeercapability);
 	newpeercapability = (peercapability | vpeercapability);