[asterisk-commits] pjsip: Add rtp timeout and rtp timeout hold endpoint options. (asterisk[13])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 24 12:09:02 CDT 2015
Mark Michelson has submitted this change and it was merged.
Change subject: pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
......................................................................
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
---
M CHANGES
M channels/chan_pjsip.c
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/26f10cadc157_add_pjsip_timeout_options.py
M include/asterisk/res_pjsip.h
M include/asterisk/res_pjsip_session.h
M include/asterisk/rtp_engine.h
M main/rtp_engine.c
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
12 files changed, 147 insertions(+), 8 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
Mark Michelson: Looks good to me, approved
Anonymous Coward #1000019: Verified
diff --git a/CHANGES b/CHANGES
index 6d30f5b..ef16ba6 100644
--- a/CHANGES
+++ b/CHANGES
@@ -44,6 +44,11 @@
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 919b926..bb79e84 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -625,6 +625,8 @@
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 6afe053..fc70947 100644
--- 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
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 cbd09e0..7c7b058 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -502,6 +502,10 @@
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 5489979..2893f66 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -79,6 +79,8 @@
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 f57f4ea..b7ac2a1 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -2304,6 +2304,22 @@
*/
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 8562558..0fca0dd 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -192,6 +192,8 @@
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 @@
return 0;
}
-
time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp)
{
return rtp->last_tx;
@@ -2204,3 +2205,13 @@
{
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 5fc6f0d..43e3da6 100644
--- 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 5d85ec8..c0c96ad 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1881,6 +1881,8 @@
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 100224c..4071ad7 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -115,10 +115,6 @@
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)) {
@@ -138,6 +134,37 @@
}
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 */
@@ -173,6 +200,8 @@
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
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 @@
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 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 9c492b3..5eb72b4 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1221,6 +1221,7 @@
}
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);
--
To view, visit https://gerrit.asterisk.org/941
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Scott Griepentrog <sgriepentrog at digium.com>
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