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If we get a codec offer using a well-known payload type, but using it for another

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codec that we don't know, Asterisk did not remove that codec from the list.

With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.

(closes issue #11376)
Reported by: lasse
Patches: 
      bug11376.txt uploaded by oej (license 306)
Tested by: lasse


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2007-11-27 15:23:17 +00:00
parent 8b650ee007
commit 0c3ec937ce
3 changed files with 52 additions and 12 deletions
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19
main/rtp.c
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@@ -1640,23 +1640,36 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
ast_mutex_unlock(&rtp->bridge_lock);
}
/*! \brief remove setting from payload type list if the rtpmap header indicates
an unknown media type */
void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt)
{
ast_mutex_lock(&rtp->bridge_lock);
rtp->current_RTP_PT[pt].isAstFormat = 0;
rtp->current_RTP_PT[pt].code = 0;
ast_mutex_unlock(&rtp->bridge_lock);
}
/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
* an SDP "a=rtpmap:" line.
* \return 0 if the MIME type was found and set, -1 if it wasn't found
*/
void ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
int ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
char *mimeType, char *mimeSubtype,
enum ast_rtp_options options)
{
unsigned int i;
int found = 0;
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
return -1; /* bogus payload type */
ast_mutex_lock(&rtp->bridge_lock);
for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
strcasecmp(mimeType, mimeTypes[i].type) == 0) {
found = 1;
rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
mimeTypes[i].payloadType.isAstFormat &&
@@ -1668,7 +1681,7 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
ast_mutex_unlock(&rtp->bridge_lock);
return;
return (found ? 0 : -1);
}
/*! \brief Return the union of all of the codecs that were set by rtp_set...() calls