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Add SIP video fixes

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2005-08-30 02:12:09 +00:00
parent 538e160956
commit 5d4d2fdfd4
8 changed files with 78 additions and 3 deletions
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rtp.c
+12 -1
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@@ -672,7 +672,7 @@ void ast_rtp_pt_default(struct ast_rtp* rtp)
rtp->rtp_lookup_code_cache_result = 0;
}
/* Make a note of a RTP payload type that was seen in a SDP "m=" line. */
/* Make a note of a RTP paymoad 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): */
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
@@ -1628,6 +1628,17 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
}
return AST_BRIDGE_COMPLETE;
} else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
(f->subclass == AST_CONTROL_VIDUPDATE)) {
ast_indicate(who == c0 ? c1 : c0, f->subclass);
ast_frfree(f);
} else {
*fo = f;
*rc = who;
ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name);
return AST_BRIDGE_COMPLETE;
}
} else {
if ((f->frametype == AST_FRAME_DTMF) ||
(f->frametype == AST_FRAME_VOICE) ||