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Merged revisions 162197 via svnmerge from

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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
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2008-12-09 19:10:33 +00:00
parent b331a9e331
commit 085cfe48cf
1 changed files with 4 additions and 4 deletions
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8
main/rtp.c
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@@ -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) */ /* 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... */ /* Somebody doesn't want to play... */
ast_channel_unlock(c0); ast_channel_unlock(c0);
if (c1) if (c1)
ast_channel_unlock(c1); ast_channel_unlock(c1);
return -1; 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); srccodec = srcpr->get_codec(c1);
else else
srccodec = 0; 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); destcodec = destpr->get_codec(c0);
else else
destcodec = 0; destcodec = 0;
@@ -2144,7 +2144,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
destcodec = 0; destcodec = 0;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ /* 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... */ /* Somebody doesn't want to play... */
ast_channel_unlock(dest); ast_channel_unlock(dest);
ast_channel_unlock(src); ast_channel_unlock(src);