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- Disable RTP hold timers while T.38 fax transmission happens

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- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
   The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
   something that video phones support in the RTP stream.
   I now this is a big architectual change at this stage for 1.4, but decided it was needed
   to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample

Issue 7679 in the bug tracker. Please test.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-12-02 11:32:51 +00:00
parent 1298cf0ea6
commit f89143bd13
4 changed files with 117 additions and 29 deletions
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15
include/asterisk/rtp.h
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@@ -219,6 +219,21 @@ struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
int ast_rtp_codec_getformat(int pt);
/*! \brief Set rtp timeout */
void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
/*! \brief Set rtp hold timeout */
void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
/*! \brief set RTP keepalive interval */
void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
/*! \brief Get RTP keepalive interval */
int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
/*! \brief Get rtp hold timeout */
int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
/*! \brief Get rtp timeout */
int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif