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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-27 19:24:17 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-27 19:24:17 +0000
commitbb9210c7a960fa9b4ec5c3d6167b25339b0428f0 (patch)
tree010456feea8c290b3ad46768771d566dba273121 /include
parent3db53911ba5ee17df139b45c6ccac18fb916d276 (diff)
The following patch with updates for trunk. Works much better in trunk.
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: http://svn.digium.com/svn/asterisk/trunk@89698 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/rtp.h8
1 files changed, 7 insertions, 1 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 56ac76e86..2aeefe077 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -178,8 +178,14 @@ void ast_rtp_pt_default(struct ast_rtp* rtp);
/*! \brief Copy payload types between RTP structures */
void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
+/*! \brief Activate payload type */
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
-void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
+
+/*! \brief clear payload type */
+void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
+
+/*! \brief Initiate payload type to a known MIME media type for a codec */
+int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
char *mimeType, char *mimeSubtype,
enum ast_rtp_options options);