From 26ab0f0fd495ae94a975ae475f618bb4459bff80 Mon Sep 17 00:00:00 2001 From: oej Date: Sat, 2 Dec 2006 11:32:51 +0000 Subject: - Disable RTP hold timers while T.38 fax transmission happens - 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: http://svn.digium.com/svn/asterisk/branches/1.4@48199 f38db490-d61c-443f-a65b-d21fe96a405b --- include/asterisk/rtp.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'include') diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index d7738b345..839648878 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 -- cgit v1.2.3