pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
[asterisk/asterisk.git] / res / res_pjsip_sdp_rtp.c
index e8654a9..125472b 100644 (file)
@@ -115,10 +115,6 @@ static int send_keepalive(const void *data)
        time_t interval;
        int send_keepalive;
 
-       if (!rtp) {
-               return 0;
-       }
-
        keepalive = ast_rtp_instance_get_keepalive(rtp);
 
        if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
@@ -140,6 +136,37 @@ static int send_keepalive(const void *data)
        return (keepalive - interval) * 1000;
 }
 
+/*! \brief Check whether RTP is being received or not */
+static int rtp_check_timeout(const void *data)
+{
+       struct ast_sip_session_media *session_media = (struct ast_sip_session_media *)data;
+       struct ast_rtp_instance *rtp = session_media->rtp;
+       int elapsed;
+       struct ast_channel *chan;
+
+       if (!rtp) {
+               return 0;
+       }
+
+       elapsed = time(NULL) - ast_rtp_instance_get_last_rx(rtp);
+       if (elapsed < ast_rtp_instance_get_timeout(rtp)) {
+               return (ast_rtp_instance_get_timeout(rtp) - elapsed) * 1000;
+       }
+
+       chan = ast_channel_get_by_name(ast_rtp_instance_get_channel_id(rtp));
+       if (!chan) {
+               return 0;
+       }
+
+       ast_log(LOG_NOTICE, "Disconnecting channel '%s' for lack of RTP activity in %d seconds\n",
+               ast_channel_name(chan), elapsed);
+
+       ast_softhangup(chan, AST_SOFTHANGUP_DEV);
+       ast_channel_unref(chan);
+
+       return 0;
+}
+
 /*! \brief Internal function which creates an RTP instance */
 static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
 {
@@ -174,6 +201,8 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
                                session->endpoint->media.cos_video, "SIP RTP Video");
        }
 
+       ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
+
        return 0;
 }
 
@@ -1272,6 +1301,28 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a
                        session_media, 1);
        }
 
+       /* As the channel lock is not held during this process the scheduled item won't block if
+        * it is hanging up the channel at the same point we are applying this negotiated SDP.
+        */
+       AST_SCHED_DEL(sched, session_media->timeout_sched_id);
+
+       /* Due to the fact that we only ever have one scheduled timeout item for when we are both
+        * off hold and on hold we don't need to store the two timeouts differently on the RTP
+        * instance itself.
+        */
+       ast_rtp_instance_set_timeout(session_media->rtp, 0);
+       if (session->endpoint->media.rtp.timeout && !session_media->remotely_held) {
+               ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout);
+       } else if (session->endpoint->media.rtp.timeout_hold && session_media->remotely_held) {
+               ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout_hold);
+       }
+
+       if (ast_rtp_instance_get_timeout(session_media->rtp)) {
+               session_media->timeout_sched_id = ast_sched_add_variable(sched,
+                       ast_rtp_instance_get_timeout(session_media->rtp) * 1000, rtp_check_timeout,
+                       session_media, 1);
+       }
+
        return 1;
 }
 
@@ -1301,9 +1352,8 @@ static void change_outgoing_sdp_stream_media_address(pjsip_tx_data *tdata, struc
 static void stream_destroy(struct ast_sip_session_media *session_media)
 {
        if (session_media->rtp) {
-               if (session_media->keepalive_sched_id != -1) {
-                       AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
-               }
+               AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
+               AST_SCHED_DEL(sched, session_media->timeout_sched_id);
                ast_rtp_instance_stop(session_media->rtp);
                ast_rtp_instance_destroy(session_media->rtp);
        }