diff options
author | Joshua Colp <jcolp@digium.com> | 2006-08-30 03:16:03 +0000 |
---|---|---|
committer | Joshua Colp <jcolp@digium.com> | 2006-08-30 03:16:03 +0000 |
commit | 12b6ec4e11f872ba59d8ffe72f479d3fd14bb00f (patch) | |
tree | 45b7b58240ee6153fe6044d3ec6d36d6c9013c93 | |
parent | da7d969ae10e89c3fb2719dbac5099e5d09c56ef (diff) |
Use an API call (ast_rtp_get_bridged) to return the RTP stream we are bridged to, and also use it in chan_sip so we know to ignore the no RTP activity checking
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r-- | channels/chan_sip.c | 15 | ||||
-rw-r--r-- | include/asterisk/rtp.h | 2 | ||||
-rw-r--r-- | main/rtp.c | 13 |
3 files changed, 20 insertions, 10 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 5508405ad..ce231bf3b 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -14365,12 +14365,15 @@ restartsearch: ast_mutex_lock(&sip->lock); } if (sip->owner) { - ast_log(LOG_NOTICE, - "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n", - sip->owner->name, - (long) (t - sip->lastrtprx)); - /* Issue a softhangup */ - ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV); + if (!(ast_rtp_get_bridged(sip->rtp))) { + ast_log(LOG_NOTICE, + "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n", + sip->owner->name, + (long) (t - sip->lastrtprx)); + /* Issue a softhangup */ + ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV); + } else + ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", sip->owner->name, (long) (t - sip->lastrtprx)); ast_channel_unlock(sip->owner); /* forget the timeouts for this call, since a hangup has already been requested and we don't want to diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index cdc81fd77..f99d4dec6 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -120,6 +120,8 @@ int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them); void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us); +struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp); + void ast_rtp_destroy(struct ast_rtp *rtp); void ast_rtp_reset(struct ast_rtp *rtp); diff --git a/main/rtp.c b/main/rtp.c index ce81738a0..2ba7ee0de 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -781,7 +781,7 @@ struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp) } /* If we are P2P bridged to another RTP stream, send it directly over */ - if (rtp->bridged && !bridge_p2p_rtcp_write(rtp, rtcpheader, res)) + if (ast_rtp_get_bridged(rtp) && !bridge_p2p_rtcp_write(rtp, rtcpheader, res)) return &ast_null_frame; if (option_debug) @@ -939,7 +939,7 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t /*! \brief Perform a Packet2Packet RTCP write */ static int bridge_p2p_rtcp_write(struct ast_rtp *rtp, unsigned int *rtcpheader, int len) { - struct ast_rtp *bridged = rtp->bridged; + struct ast_rtp *bridged = ast_rtp_get_bridged(rtp); int res = 0; /* If RTCP is not present on the bridged RTP session, then ignore this */ @@ -962,7 +962,7 @@ static int bridge_p2p_rtcp_write(struct ast_rtp *rtp, unsigned int *rtcpheader, /*! \brief Perform a Packet2Packet RTP write */ static int bridge_p2p_rtp_write(struct ast_rtp *rtp, unsigned int *rtpheader, int len, int hdrlen) { - struct ast_rtp *bridged = rtp->bridged; + struct ast_rtp *bridged = ast_rtp_get_bridged(rtp); int res = 0, payload = 0, bridged_payload = 0, version, padding, mark, ext; struct rtpPayloadType rtpPT; unsigned int seqno; @@ -1084,7 +1084,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp) } /* If we are bridged to another RTP stream, send direct */ - if (rtp->bridged && !bridge_p2p_rtp_write(rtp, rtpheader, res, hdrlen)) + if (ast_rtp_get_bridged(rtp) && !bridge_p2p_rtp_write(rtp, rtpheader, res, hdrlen)) return &ast_null_frame; if (version != 2) @@ -1846,6 +1846,11 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us) *us = rtp->us; } +struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp) +{ + return rtp->bridged; +} + void ast_rtp_stop(struct ast_rtp *rtp) { if (rtp->rtcp && rtp->rtcp->schedid > 0) { |