diff options
author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-09 19:06:14 +0000 |
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committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-09 19:06:14 +0000 |
commit | 5887d5222d7837dd980d38d26defe6b274a22e62 (patch) | |
tree | 219c934369c668d611ae1316101f77846d96b884 /main/rtp.c | |
parent | da16f32ce784f7ad469714e646649cb6beb8ad4b (diff) |
Take video into account when early bridging RTP.
(closes issue #13535)
Reported by: davidw
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@162188 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 fa9da999a..2cdfeedc1 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -1524,18 +1524,18 @@ int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src) } /* 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 @@ 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); |