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

diff --git a/CHANGES b/CHANGES
index 6d30f5be600065ad3be7bfe4efc2552c38d0aecb..ef16ba65d8d7730f2c9e3f651a74d7d3f91426f1 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -44,6 +44,11 @@ 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 ------------
 ------------------------------------------------------------------------------

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 919b926184eb0ae3fa703304f332408d401afd8d..bb79e84a50578fa0a3f8c234b42bbecac532d338 100644 (file)
--- a/channels/chan_pjsip.c
+++ b/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;
        }

diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 6afe053c3885732c693c0951d7262dfffc36c38f..fc70947568301ae0695bb95bcf107f95fbd7c640 100644 (file)
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -735,6 +735,12 @@
 ;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]

diff --git a/contrib/ast-db-manage/config/versions/26f10cadc157_add_pjsip_timeout_options.py b/contrib/ast-db-manage/config/versions/26f10cadc157_add_pjsip_timeout_options.py
new file mode 100644 (file)
index 0000000..8972d80
--- /dev/null
+++ b/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')

diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index cbd09e0e0b310b565697f1a16d800828a6651c22..7c7b058cfddc4210bbd9463822837fedab53560b 100644 (file)
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -502,6 +502,10 @@ 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;
 };
 
 /*!

diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 5489979ed7db4a5569745240840ef60957c44d65..2893f66ab8d6d9d59c1c9db6c3ed29b8467bb799 100644 (file)
--- a/include/asterisk/res_pjsip_session.h
+++ b/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 */
        unsigned int held:1;
        /*! \brief Stream type this session media handles */

diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index f57f4ea35a61bb6c042f127e11efce18de5e6824..b7ac2a149ef8a3aeb0c07d34ef955b97c37c41b0 100644 (file)
--- a/include/asterisk/rtp_engine.h
+++ b/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
  * @{
  */

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 8562558b818ed2f4909683ee43afc6afa4353c1b..0fca0dded48b1f6f8d74e098db2dc9b556d02b88 100644 (file)
--- a/main/rtp_engine.c
+++ b/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 */
@@ -2194,7 +2196,6 @@ 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;
@@ -2204,3 +2205,13 @@ 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;
+}

diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 5fc6f0d230d051f94b3e133934b492e15505e8a0..43e3da6ded056bebe3fc7654dbd4c395b0787c20 100644 (file)
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -795,6 +795,22 @@
                                                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>

diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 5d85ec880338cb79ac731ef5bdccb07d619c2f09..c0c96ad7feaba40cb715c9f19555d96477476b4f 100644 (file)
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1881,6 +1881,8 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
        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);

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 100224c10e27b2f9ffacb8e62250ec2930535bf7..4071ad76a48756c88ee48f670b255b775d3f9179 100644 (file)
--- a/res/res_pjsip_sdp_rtp.c
+++ b/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;
 }
 
@@ -1271,6 +1300,28 @@ 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->held) {
+               ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout);
+       } else if (session->endpoint->media.rtp.timeout_hold && session_media->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;
 }
 
@@ -1300,9 +1351,8 @@ 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);
        }

diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 9c492b3fcb39d5b9c2585738cf317581fac5f53e..5eb72b45afd90e36e0e5858fe4490a6bff418828 100644 (file)
--- a/res/res_pjsip_session.c
+++ b/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);