diff options
| -rw-r--r-- | CHANGES | 5 | ||||
| -rw-r--r-- | channels/chan_pjsip.c | 2 | ||||
| -rw-r--r-- | configs/samples/pjsip.conf.sample | 6 | ||||
| -rw-r--r-- | contrib/ast-db-manage/config/versions/26f10cadc157_add_pjsip_timeout_options.py | 24 | ||||
| -rw-r--r-- | include/asterisk/res_pjsip.h | 4 | ||||
| -rw-r--r-- | include/asterisk/res_pjsip_session.h | 2 | ||||
| -rw-r--r-- | include/asterisk/rtp_engine.h | 16 | ||||
| -rw-r--r-- | main/rtp_engine.c | 13 | ||||
| -rw-r--r-- | res/res_pjsip.c | 16 | ||||
| -rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 2 | ||||
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 64 | ||||
| -rw-r--r-- | res/res_pjsip_session.c | 1 |
12 files changed, 147 insertions, 8 deletions
@@ -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 919b92618..bb79e84a5 100644 --- 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 6afe053c3..fc7094756 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 000000000..8972d8030 --- /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 cbd09e0e0..7c7b058cf 100644 --- 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 5489979ed..2893f66ab 100644 --- 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 f57f4ea35..b7ac2a149 100644 --- 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 8562558b8..0fca0dded 100644 --- 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 5fc6f0d23..43e3da6de 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 5d85ec880..c0c96ad7f 100644 --- 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 100224c10..4071ad76a 100644 --- 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 9c492b3fc..5eb72b45a 100644 --- 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); |