diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-09 19:08:39 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-09 19:08:39 +0000 |
| commit | 274f1bd4becfcbd4fb11520450e7796a04916023 (patch) | |
| tree | a2ee8a0d52afcc557ec3bed73d3e8a36275a28eb /main/rtp.c | |
| parent | 7ef05745e75744937494cff9112d17c5da20c191 (diff) | |
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: http://svn.digium.com/svn/asterisk/trunk@162197 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/main/rtp.c b/main/rtp.c index 9f4c34284..1cf482e05 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); |