[asterisk-commits] file: trunk r38030 - /trunk/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: file
Date: Thu Jul 20 13:42:22 2006
New Revision: 38030

URL: http://svn.digium.com/view/asterisk?rev=38030&view=rev
Log:
Only bitmaskify the RTP payload structure for video if an RTP structure exists for it... otherwise the default values will cause codec combination madness

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=38030&r1=38029&r2=38030&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Jul 20 13:42:22 2006
@@ -4845,7 +4845,8 @@
 
 	/* 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);