Use an API call (ast_rtp_get_bridged) to return the RTP stream we are bridged to...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5508405..ce231bf 100644 (file)
--- 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 cdc81fd..f99d4de 100644 (file)
--- 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 ce81738..2ba7ee0 100644 (file)
--- 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) {