diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-06 16:09:33 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-06 16:09:33 +0000 |
commit | 4506e03f3d24f4bd4f6b17803031ede1e353de28 (patch) | |
tree | b849a5c247c05d8e052b854defc2526e0f341a64 /rtp.c | |
parent | 77abfdfd00db4460aeb6e9c3b5065406486ca4f4 (diff) |
Merge of the "sdpcleanup" branch. Thanks to John Martin for a lot of tests
and some patches (all disclaimed).
- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments
Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32597 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'rtp.c')
-rw-r--r-- | rtp.c | 7 |
1 files changed, 3 insertions, 4 deletions
@@ -1350,7 +1350,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i return 1; } -/*! \brief Make a note of a RTP paymoad type that was seen in a SDP "m=" line. +/*! \brief Make a note of a RTP payload type that was seen in a SDP "m=" line. * By default, use the well-known value for this type (although it may * still be set to a different value by a subsequent "a=rtpmap:" line) */ @@ -1359,9 +1359,8 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) if (pt < 0 || pt > MAX_RTP_PT) return; /* bogus payload type */ - if (static_RTP_PT[pt].code != 0) { + if (static_RTP_PT[pt].code != 0) rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; - } } /*! \brief Make a note of a RTP payload type (with MIME type) that was seen in @@ -2245,7 +2244,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f) /* Make sure we have enough space for RTP header */ if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { - ast_log(LOG_WARNING, "RTP can only send voice\n"); + ast_log(LOG_WARNING, "RTP can only send voice and video\n"); return -1; } |