diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-03-08 18:05:54 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-03-08 18:05:54 +0000 |
| commit | e676dc69e54b933fa05d8277d03c5e961daceca1 (patch) | |
| tree | 82b0235722534aca492e9ed5c7b7c8e72c840858 /main/rtp.c | |
| parent | 21e626b1e7db99c69c579ca432952c36897069b6 (diff) | |
Merged revisions 58436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines
Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58437 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 18 |
1 files changed, 12 insertions, 6 deletions
diff --git a/main/rtp.c b/main/rtp.c index 1ff0a453d..a6a473a5a 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -1627,7 +1627,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED; enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED; - int srccodec; + int srccodec, destcodec; /* Lock channels */ ast_channel_lock(dest); @@ -1661,8 +1661,18 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED; text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(src, &tsrcp) : AST_RTP_GET_FAILED; + /* Ensure we have at least one matching codec */ + if (srcpr->get_codec) + srccodec = srcpr->get_codec(src); + else + srccodec = 0; + if (destpr->get_codec) + destcodec = destpr->get_codec(dest); + else + 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) { + if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) { /* Somebody doesn't want to play... */ ast_channel_unlock(dest); ast_channel_unlock(src); @@ -1673,10 +1683,6 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i ast_rtp_pt_copy(vdestp, vsrcp); if (tdestp && tsrcp) ast_rtp_pt_copy(tdestp, tsrcp); - if (srcpr->get_codec) - srccodec = srcpr->get_codec(src); - else - srccodec = 0; if (media) { /* Bridge early */ if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) |