From: Joshua Colp <jcolp@digium.com>
Date: Thu, 20 Jul 2006 18:42:22 +0000 (+0000)
Subject: Only bitmaskify the RTP payload structure for video if an RTP structure exists for... 
X-Git-Tag: 1.4.0-beta1~542
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=1861dcdff5666cbac27288f7bdfdf40104d29d4d;p=thirdparty%2Fasterisk.git

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
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a3959c8de5..d621e72186 100644
--- a/channels/chan_sip.c
+++ b/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);