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

1 files changed, 3 insertions, 4 deletions

diff --git a/rtp.c b/rtp.c
index a50350349..1e806f74d 100644
--- a/rtp.c
+++ b/rtp.c
@@ -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;
 	}