diff options
| author | Olle Johansson <oej@edvina.net> | 2006-12-02 12:05:40 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2006-12-02 12:05:40 +0000 |
| commit | c23bc46089970c9c7275b662bbd48b0ed7310fc6 (patch) | |
| tree | 09ffb0a36d6c9a88e86937e5c2cff44de6ff208f /include | |
| parent | eef9f7958bea41d443fb4e0698636459ab28446a (diff) | |
- Disable RTP timeouts during T.38 transmission
- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include')
| -rw-r--r-- | include/asterisk/rtp.h | 15 |
1 files changed, 15 insertions, 0 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index aab598a31..f7c1718f9 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -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 |