From 085cfe48cf5ef20370ab0cf6d97669c7a6b6b2c9 Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Tue, 9 Dec 2008 19:10:33 +0000 Subject: [PATCH] Merged revisions 162197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162197 | file | 2008-12-09 15:08:39 -0400 (Tue, 09 Dec 2008) | 11 lines 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- main/rtp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/main/rtp.c b/main/rtp.c index aac30de490..6af9b48863 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -2056,18 +2056,18 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1) } /* 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 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i 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);