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Merged revisions 58436 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.4

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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: https://origsvn.digium.com/svn/asterisk/trunk@58437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2007-03-08 18:05:54 +00:00
parent 6dc98369bf
commit 2ab6ed30cd
1 changed files with 12 additions and 6 deletions
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18
main/rtp.c
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@@ -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; 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_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; 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 */ /* Lock channels */
ast_channel_lock(dest); 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; 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; 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) */ /* 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... */ /* Somebody doesn't want to play... */
ast_channel_unlock(dest); ast_channel_unlock(dest);
ast_channel_unlock(src); 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); ast_rtp_pt_copy(vdestp, vsrcp);
if (tdestp && tsrcp) if (tdestp && tsrcp)
ast_rtp_pt_copy(tdestp, tsrcp); ast_rtp_pt_copy(tdestp, tsrcp);
if (srcpr->get_codec)
srccodec = srcpr->get_codec(src);
else
srccodec = 0;
if (media) { if (media) {
/* Bridge early */ /* Bridge early */
if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE))) if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))