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https://github.com/asterisk/asterisk.git
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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
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@@ -14365,12 +14365,15 @@ restartsearch:
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ast_mutex_lock(&sip->lock);
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}
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if (sip->owner) {
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ast_log(LOG_NOTICE,
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"Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
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sip->owner->name,
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(long) (t - sip->lastrtprx));
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/* Issue a softhangup */
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ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV);
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if (!(ast_rtp_get_bridged(sip->rtp))) {
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ast_log(LOG_NOTICE,
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"Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
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sip->owner->name,
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(long) (t - sip->lastrtprx));
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/* Issue a softhangup */
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ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV);
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} else
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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));
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ast_channel_unlock(sip->owner);
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/* forget the timeouts for this call, since a hangup
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has already been requested and we don't want to
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@@ -120,6 +120,8 @@ int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
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void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
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struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
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void ast_rtp_destroy(struct ast_rtp *rtp);
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void ast_rtp_reset(struct ast_rtp *rtp);
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13
main/rtp.c
13
main/rtp.c
@@ -781,7 +781,7 @@ struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp)
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}
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/* If we are P2P bridged to another RTP stream, send it directly over */
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if (rtp->bridged && !bridge_p2p_rtcp_write(rtp, rtcpheader, res))
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if (ast_rtp_get_bridged(rtp) && !bridge_p2p_rtcp_write(rtp, rtcpheader, res))
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return &ast_null_frame;
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if (option_debug)
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@@ -939,7 +939,7 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
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/*! \brief Perform a Packet2Packet RTCP write */
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static int bridge_p2p_rtcp_write(struct ast_rtp *rtp, unsigned int *rtcpheader, int len)
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{
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struct ast_rtp *bridged = rtp->bridged;
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struct ast_rtp *bridged = ast_rtp_get_bridged(rtp);
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int res = 0;
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/* If RTCP is not present on the bridged RTP session, then ignore this */
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@@ -962,7 +962,7 @@ static int bridge_p2p_rtcp_write(struct ast_rtp *rtp, unsigned int *rtcpheader,
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/*! \brief Perform a Packet2Packet RTP write */
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static int bridge_p2p_rtp_write(struct ast_rtp *rtp, unsigned int *rtpheader, int len, int hdrlen)
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{
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struct ast_rtp *bridged = rtp->bridged;
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struct ast_rtp *bridged = ast_rtp_get_bridged(rtp);
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int res = 0, payload = 0, bridged_payload = 0, version, padding, mark, ext;
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struct rtpPayloadType rtpPT;
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unsigned int seqno;
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@@ -1084,7 +1084,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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}
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/* If we are bridged to another RTP stream, send direct */
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if (rtp->bridged && !bridge_p2p_rtp_write(rtp, rtpheader, res, hdrlen))
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if (ast_rtp_get_bridged(rtp) && !bridge_p2p_rtp_write(rtp, rtpheader, res, hdrlen))
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return &ast_null_frame;
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if (version != 2)
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@@ -1846,6 +1846,11 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
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*us = rtp->us;
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}
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struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp)
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{
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return rtp->bridged;
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}
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void ast_rtp_stop(struct ast_rtp *rtp)
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{
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if (rtp->rtcp && rtp->rtcp->schedid > 0) {
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