kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-24 05:38:11 +00:00
Code Issues Packages Projects Releases Wiki Activity

The following patch with updates for trunk. Works much better in trunk.

Browse Source 
Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...

Merged revisions 89630 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines

If we get a codec offer using a well-known payload type, but using it for another
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/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2007-11-27 19:24:17 +00:00
parent 35ecd08b46
commit df7ba90b20
3 changed files with 45 additions and 12 deletions
Download Patch File Download Diff File Expand all files Collapse all files

19
main/rtp.c
View File

@@ -1939,23 +1939,36 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
rtp_bridge_unlock(rtp);
}
/*! \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)
{
rtp_bridge_lock(rtp);
rtp->current_RTP_PT[pt].isAstFormat = 0;
rtp->current_RTP_PT[pt].code = 0;
rtp_bridge_unlock(rtp);
}
/*! \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 */
rtp_bridge_lock(rtp);
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 &&
@@ -1967,7 +1980,7 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
rtp_bridge_unlock(rtp);
return;
return (found ? 0 : -1);
}
/*! \brief Return the union of all of the codecs that were set by rtp_set...() calls