From 12b6ec4e11f872ba59d8ffe72f479d3fd14bb00f Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Wed, 30 Aug 2006 03:16:03 +0000
Subject: [PATCH] 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
---
 channels/chan_sip.c    | 15 +++++++++------
 include/asterisk/rtp.h |  2 ++
 main/rtp.c             | 13 +++++++++----
 3 files changed, 20 insertions(+), 10 deletions(-)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5508405ad5..ce231bf3b0 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 cdc81fd772..f99d4dec63 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 ce81738a0d..2ba7ee0de4 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) {