[asterisk-commits] oej: branch oej/videocaps r103758 - /team/oej/videocaps/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Mon Feb 18 07:15:43 2008
New Revision: 103758

URL: http://svn.digium.com/view/asterisk?view=rev&rev=103758
Log:
Make sure the bandwidth header is directly after the video media session declaration.

Modified:
    team/oej/videocaps/channels/chan_sip.c

Modified: team/oej/videocaps/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_sip.c?view=diff&rev=103758&r1=103757&r2=103758
==============================================================================
--- team/oej/videocaps/channels/chan_sip.c (original)
+++ team/oej/videocaps/channels/chan_sip.c Mon Feb 18 07:15:43 2008
@@ -8540,6 +8540,10 @@
 	else
 		hold = "a=sendrecv\r\n";
 
+	/* The bandwidth selector for video needs to be first in the video media session declaration */
+ 	if (p->jointcaps.maxvideobitrate > 0)
+		ast_str_append(&a_video, 0, "b=AS:%d\r\n", p->jointcaps.maxvideobitrate/1000);
+
 	/* Now, start adding audio codecs. These are added in this order:
 		- First what was requested by the calling channel
 		- Then preferences in order from sip.conf device config for this peer/user
@@ -8626,8 +8630,6 @@
 			m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 ||
 			a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2)
 		ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
- 	if (p->jointcaps.maxvideobitrate > 0)
-		ast_str_append(&a_video, 0, "b=AS:%d\r\n", p->jointcaps.maxvideobitrate/1000);
 
  	ast_str_append(&m_audio, 0, "\r\n");
  	if (needvideo)