diff options
| author | Joshua Colp <jcolp@digium.com> | 2006-12-26 04:34:07 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2006-12-26 04:34:07 +0000 |
| commit | 7f61b822c17ccadac726172a2b120e8c9d029abf (patch) | |
| tree | 756df2c8bb71afc320d31e4d30afe941fb099ef6 /include/asterisk/rtp.h | |
| parent | b3ab5300776cb22075d6add23ec27d6a968e0f5c (diff) | |
Merged revisions 48964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include/asterisk/rtp.h')
| -rw-r--r-- | include/asterisk/rtp.h | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index a93f39261..2edff8cad 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -223,6 +223,7 @@ int ast_rtcp_send_h261fur(void *data); char *ast_rtp_get_quality(struct ast_rtp *rtp); /*! \brief Return RTCP quality string */ void ast_rtp_init(void); /*! Initialize RTP subsystem */ int ast_rtp_reload(void); /*! reload rtp configuration */ +void ast_rtp_new_init(struct ast_rtp *rtp); /*! Set codec preference */ int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs); |