[asterisk-commits] file: trunk r162197 - in /trunk: ./ main/rtp.c

[SVN commits to the Asterisk project]

Author: file
Date: Tue Dec  9 13:08:39 2008
New Revision: 162197

URL: http://svn.digium.com/view/asterisk?view=rev&rev=162197
Log:
Merged revisions 162188 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines
  
  Take video into account when early bridging RTP.
  (closes issue #13535)
  Reported by: davidw
........

Modified:
    trunk/   (props changed)
    trunk/main/rtp.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=162197&r1=162196&r2=162197
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Tue Dec  9 13:08:39 2008
@@ -2056,18 +2056,18 @@
 	}
 
 	/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
-	if (audio_dest_res != AST_RTP_TRY_NATIVE) {
+	if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {
 		/* Somebody doesn't want to play... */
 		ast_channel_unlock(c0);
 		if (c1)
 			ast_channel_unlock(c1);
 		return -1;
 	}
-	if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
+	if (audio_src_res == AST_RTP_TRY_NATIVE && (video_src_res == AST_RTP_GET_FAILED || video_src_res == AST_RTP_TRY_NATIVE) && srcpr->get_codec)
 		srccodec = srcpr->get_codec(c1);
 	else
 		srccodec = 0;
-	if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
+	if (audio_dest_res == AST_RTP_TRY_NATIVE && (video_dest_res == AST_RTP_GET_FAILED || video_dest_res == AST_RTP_TRY_NATIVE) && destpr->get_codec)
 		destcodec = destpr->get_codec(c0);
 	else
 		destcodec = 0;
@@ -2144,7 +2144,7 @@
 		destcodec = 0;
 
 	/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
-	if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
+	if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) {
 		/* Somebody doesn't want to play... */
 		ast_channel_unlock(dest);
 		ast_channel_unlock(src);