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Merged revisions 48964 via svnmerge from

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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
This commit is contained in:
Joshua Colp
2006-12-26 04:34:07 +00:00
parent b3ab530077
commit 7f61b822c1
3 changed files with 21 additions and 6 deletions
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include/asterisk/rtp.h
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@@ -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);