pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.

This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.

ASTERISK-25259 #close

Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9

CHANGES

@@ -213,6 +213,11 @@ res_pjsip
   an interval, in seconds, at which we will send RTP comfort noise packets to
   the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
 
+* New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
+  These options specify the amount of time, in seconds, that Asterisk will wait
+  before terminating the call due to lack of received RTP. These are identical
+  to chan_sip's rtptimeout and rtpholdtimeout options.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
 ------------------------------------------------------------------------------

channels/chan_pjsip.c

@@ -625,6 +625,8 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
 		return f;
 	}
 
+	ast_rtp_instance_set_last_rx(media->rtp, time(NULL));
+
 	if (f->frametype != AST_FRAME_VOICE) {
 		return f;
 	}

configs/samples/pjsip.conf.sample

@@ -737,6 +737,12 @@
 ;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if
                 ; RTP is not flowing. This setting is useful for ensuring that
                 ; holes in NATs and firewalls are kept open throughout a call.
+;rtp_timeout=      ; Hang up channel if RTP is not received for the specified
+                   ; number of seconds when the channel is off hold (default:
+                   ; "0" or not enabled)
+;rtp_timeout_hold= ; Hang up channel if RTP is not received for the specified
+                   ; number of seconds when the channel is on hold (default:
+                   ; "0" or not enabled)
 
 ;==========================AUTH SECTION OPTIONS=========================
 ;[auth]

contrib/ast-db-manage/config/versions/26f10cadc157_add_pjsip_timeout_options.py

@@ -0,0 +1,24 @@
+"""add pjsip timeout options
+
+Revision ID: 26f10cadc157
+Revises: 498357a710ae
+Create Date: 2015-07-21 07:45:00.696965
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '26f10cadc157'
+down_revision = '498357a710ae'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def upgrade():
+    op.add_column('ps_endpoints', sa.Column('rtp_timeout', sa.Integer))
+    op.add_column('ps_endpoints', sa.Column('rtp_timeout_hold', sa.Integer))
+
+
+def downgrade():
+    op.drop_column('ps_endpoints', 'rtp_timeout')
+    op.drop_column('ps_endpoints', 'rtp_timeout_hold')

include/asterisk/res_pjsip.h

@@ -504,6 +504,10 @@ struct ast_sip_media_rtp_configuration {
 	unsigned int encryption_optimistic;
 	/*! Number of seconds between RTP keepalive packets */
 	unsigned int keepalive;
+	/*! Number of seconds before terminating channel due to lack of RTP (when not on hold) */
+	unsigned int timeout;
+	/*! Number of seconds before terminating channel due to lack of RTP (when on hold) */
+	unsigned int timeout_hold;
 };
 
 /*!

include/asterisk/res_pjsip_session.h

@@ -79,6 +79,8 @@ struct ast_sip_session_media {
 	pj_str_t transport;
 	/*! \brief Scheduler ID for RTP keepalive */
 	int keepalive_sched_id;
+	/*! \brief Scheduler ID for RTP timeout */
+	int timeout_sched_id;
 	/*! \brief Stream is on hold by remote side */
 	unsigned int remotely_held:1;
 	/*! \brief Stream is on hold by local side */

include/asterisk/rtp_engine.h

@@ -2304,6 +2304,22 @@ time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp);
  */
 void ast_rtp_instance_set_last_tx(struct ast_rtp_instance *rtp, time_t time);
 
+/*
+ * \brief Get the last RTP reception time
+ *
+ * \param rtp The instance from which to get the last reception time
+ * \return The last RTP reception time
+ */
+time_t ast_rtp_instance_get_last_rx(const struct ast_rtp_instance *rtp);
+
+/*!
+ * \brief Set the last RTP reception time
+ *
+ * \param rtp The instance on which to set the last reception time
+ * \param time The last reception time
+ */
+void ast_rtp_instance_set_last_rx(struct ast_rtp_instance *rtp, time_t time);
+
 /*! \addtogroup StasisTopicsAndMessages
  * @{
  */

main/rtp_engine.c

@@ -192,6 +192,8 @@ struct ast_rtp_instance {
 	char channel_uniqueid[AST_MAX_UNIQUEID];
 	/*! Time of last packet sent */
 	time_t last_tx;
+	/*! Time of last packet received */
+	time_t last_rx;
 };
 
 /*! List of RTP engines that are currently registered */
@@ -2209,7 +2211,6 @@ int ast_rtp_engine_init()
 	return 0;
 }
 
-
 time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp)
 {
 	return rtp->last_tx;
@@ -2219,3 +2220,13 @@ void ast_rtp_instance_set_last_tx(struct ast_rtp_instance *rtp, time_t time)
 {
 	rtp->last_tx = time;
 }
+
+time_t ast_rtp_instance_get_last_rx(const struct ast_rtp_instance *rtp)
+{
+	return rtp->last_rx;
+}
+
+void ast_rtp_instance_set_last_rx(struct ast_rtp_instance *rtp, time_t time)
+{
+	rtp->last_rx = time;
+}

res/res_pjsip.c

@@ -798,6 +798,22 @@
 						a hole open in order to allow for media to arrive at Asterisk.
 					</para></description>
 				</configOption>
+				<configOption name="rtp_timeout" default="0">
+					<synopsis>Maximum number of seconds without receiving RTP (while off hold) before terminating call.</synopsis>
+					<description><para>
+						This option configures the number of seconds without RTP (while off hold) before
+						considering a channel as dead. When the number of seconds is reached the underlying
+						channel is hung up. By default this option is set to 0, which means do not check.
+					</para></description>
+				</configOption>
+				<configOption name="rtp_timeout_hold" default="0">
+					<synopsis>Maximum number of seconds without receiving RTP (while on hold) before terminating call.</synopsis>
+					<description><para>
+						This option configures the number of seconds without RTP (while on hold) before
+						considering a channel as dead. When the number of seconds is reached the underlying
+						channel is hung up. By default this option is set to 0, which means do not check.
+					</para></description>
+				</configOption>
 			</configObject>
 			<configObject name="auth">
 				<synopsis>Authentication type</synopsis>

res/res_pjsip/pjsip_configuration.c

@@ -1881,6 +1881,8 @@ int ast_res_pjsip_initialize_configuration(void)
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "force_avp", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.force_avp));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "media_use_received_transport", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.use_received_transport));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtp_keepalive", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.rtp.keepalive));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtp_timeout", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.rtp.timeout));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtp_timeout_hold", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.rtp.timeout_hold));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "one_touch_recording", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, info.recording.enabled));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "inband_progress", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, inband_progress));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "call_group", "", group_handler, callgroup_to_str, NULL, 0, 0);

res/res_pjsip_sdp_rtp.c

@@ -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);
 	}

res/res_pjsip_session.c

@@ -1221,6 +1221,7 @@ static int add_session_media(void *obj, void *arg, int flags)
 	}
 	session_media->encryption = session->endpoint->media.rtp.encryption;
 	session_media->keepalive_sched_id = -1;
+	session_media->timeout_sched_id = -1;
 	/* Safe use of strcpy */
 	strcpy(session_media->stream_type, handler_list->stream_type);
 	ao2_link(session->media, session_media);