[asterisk-commits] file: branch 1.4 r162188 - /branches/1.4/main/rtp.c

[SVN commits to the Asterisk project]

Author: file
Date: Tue Dec  9 13:06:14 2008
New Revision: 162188

URL: http://svn.digium.com/view/asterisk?view=rev&rev=162188
Log:
Take video into account when early bridging RTP.
(closes issue #13535)
Reported by: davidw

Modified:
    branches/1.4/main/rtp.c

Modified: branches/1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/rtp.c?view=diff&rev=162188&r1=162187&r2=162188
==============================================================================
--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Tue Dec  9 13:06:14 2008
@@ -1524,18 +1524,18 @@
 	}
 
 	/* 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(dest);
 		if (src)
 			ast_channel_unlock(src);
 		return 0;
 	}
-	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(src);
 	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(dest);
 	else
 		destcodec = 0;
@@ -1613,7 +1613,7 @@
 		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);