From: Joshua Colp Date: Mon, 3 Nov 2014 14:45:01 +0000 (+0000) Subject: chan_pjsip: Add support for passing hold and unhold requests through. X-Git-Tag: 13.30.0-rc1~57^2 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=d775bc5addbde89f3e4baa03884944c8c4cc2aa4;p=thirdparty%2Fasterisk.git chan_pjsip: Add support for passing hold and unhold requests through. This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ Change-Id: Ib6294e906e577e1a4245cb1f058d3976ff484c52 --- diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c index b0d5fda5d9..2a243b6d10 100644 --- a/channels/chan_pjsip.c +++ b/channels/chan_pjsip.c @@ -1349,6 +1349,39 @@ static int update_connected_line_information(void *data) return 0; } +/*! \brief Callback which changes the value of locally held on the media stream */ +static int local_hold_set_state(void *obj, void *arg, int flags) +{ + struct ast_sip_session_media *session_media = obj; + unsigned int *held = arg; + + session_media->locally_held = *held; + + return 0; +} + +/*! \brief Update local hold state and send a re-INVITE with the new SDP */ +static int remote_send_hold_refresh(struct ast_sip_session *session, unsigned int held) +{ + ao2_callback(session->media, OBJ_NODATA, local_hold_set_state, &held); + ast_sip_session_refresh(session, NULL, NULL, NULL, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1); + ao2_ref(session, -1); + + return 0; +} + +/*! \brief Update local hold state to be held */ +static int remote_send_hold(void *data) +{ + return remote_send_hold_refresh(data, 1); +} + +/*! \brief Update local hold state to be unheld */ +static int remote_send_unhold(void *data) +{ + return remote_send_hold_refresh(data, 0); +} + /*! \brief Function called by core to ask the channel to indicate some sort of condition */ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen) { @@ -1493,7 +1526,15 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi device_buf = alloca(device_buf_size); ast_channel_get_device_name(ast, device_buf, device_buf_size); ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf); - ast_moh_start(ast, data, NULL); + if (!channel->session->endpoint->moh_passthrough) { + ast_moh_start(ast, data, NULL); + } else { + if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) { + ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n", + ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint)); + ao2_ref(channel->session, -1); + } + } break; case AST_CONTROL_UNHOLD: chan_pjsip_remove_hold(ast_channel_uniqueid(ast)); @@ -1501,7 +1542,15 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi device_buf = alloca(device_buf_size); ast_channel_get_device_name(ast, device_buf, device_buf_size); ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf); - ast_moh_stop(ast); + if (!channel->session->endpoint->moh_passthrough) { + ast_moh_stop(ast); + } else { + if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) { + ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n", + ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint)); + ao2_ref(channel->session, -1); + } + } break; case AST_CONTROL_SRCUPDATE: break; diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c index 76f351f1e8..6cdc7213b3 100644 --- a/channels/pjsip/dialplan_functions.c +++ b/channels/pjsip/dialplan_functions.c @@ -592,7 +592,7 @@ static int channel_read_rtp(struct ast_channel *chan, const char *type, const ch snprintf(buf, buflen, "%d", 0); } } else if (!strcmp(type, "hold")) { - snprintf(buf, buflen, "%d", media->held ? 1 : 0); + snprintf(buf, buflen, "%d", media->remotely_held ? 1 : 0); } else { ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type); return -1; diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample index bbd0317e79..9e6365e253 100644 --- a/configs/samples/pjsip.conf.sample +++ b/configs/samples/pjsip.conf.sample @@ -655,6 +655,8 @@ ; An MWI subscribe will replace unsoliticed NOTIFYs ; (default: "no") ;moh_suggest=default ; Default Music On Hold class (default: "default") +;moh_passthrough=yes ; Pass Music On Hold through using SIP re-invites with sendonly + ; when placing on hold and sendrecv when taking off hold ;outbound_auth= ; Authentication object used for outbound requests (default: ; "") ;outbound_proxy= ; Proxy through which to send requests, a full SIP URI diff --git a/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py b/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py new file mode 100644 index 0000000000..cf7baf64a5 --- /dev/null +++ b/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py @@ -0,0 +1,30 @@ +"""Add moh_passthrough option to pjsip + +Revision ID: 339e1dfa644d +Revises: 3a094a18e75b +Create Date: 2014-10-21 14:55:34.197448 + +""" + +# revision identifiers, used by Alembic. +revision = '339e1dfa644d' +down_revision = '3a094a18e75b' + +from alembic import op +import sqlalchemy as sa +from sqlalchemy.dialects.postgresql import ENUM + +YESNO_NAME = 'yesno_values' +YESNO_VALUES = ['yes', 'no'] + +def upgrade(): + ############################# Enums ############################## + + # yesno_values have already been created, so use postgres enum object + # type to get around "already created" issue - works okay with mysql + yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False) + + op.add_column('ps_endpoints', sa.Column('moh_passthrough', yesno_values)) + +def downgrade(): + op.drop_column('ps_endpoints', 'moh_passthrough') diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h index 5e8dd3a24f..163059531c 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -802,6 +802,8 @@ struct ast_sip_endpoint { unsigned int send_connected_line; /*! Ignore 183 if no SDP is present */ unsigned int ignore_183_without_sdp; + /*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */ + unsigned int moh_passthrough; }; /*! URI parameter for symmetric transport */ diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h index e2d4fcd27f..51d75c6d6d 100644 --- a/include/asterisk/res_pjsip_session.h +++ b/include/asterisk/res_pjsip_session.h @@ -81,8 +81,10 @@ struct ast_sip_session_media { 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 is on hold by remote side */ + unsigned int remotely_held:1; + /*! \brief Stream is on hold by local side */ + unsigned int locally_held:1; /*! \brief Does remote support rtcp_mux */ unsigned int remote_rtcp_mux:1; /*! \brief Does remote support ice */ diff --git a/res/res_pjsip.c b/res/res_pjsip.c index c41b545b49..ec159d6fa5 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -744,6 +744,9 @@ Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number + + Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side + String placed as the username portion of an SDP origin (o=) line. @@ -2286,6 +2289,9 @@ + + + diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index b5525beeb2..db53592588 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1858,6 +1858,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod ast_sorcery_object_field_register(sip_sorcery, "endpoint", "record_off_feature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, info.recording.offfeature)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allow_transfer", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allowtransfer)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "user_eq_phone", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, usereqphone)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "moh_passthrough", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, moh_passthrough)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_owner", "-", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpowner)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_session", "Asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpsession)); ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "tos_audio", "0", tos_handler, tos_audio_to_str, NULL, 0, 0); diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 76d47a3f6d..98e981965f 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -1184,6 +1184,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as static const pj_str_t STR_IP4 = { "IP4", 3}; static const pj_str_t STR_IP6 = { "IP6", 3}; static const pj_str_t STR_SENDRECV = { "sendrecv", 8 }; + static const pj_str_t STR_SENDONLY = { "sendonly", 8 }; pjmedia_sdp_media *media; const char *hostip = NULL; struct ast_sockaddr addr; @@ -1369,7 +1370,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as /* Add the sendrecv attribute - we purposely don't keep track because pjmedia-sdp will automatically change our offer for us */ attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr); - attr->name = STR_SENDRECV; + attr->name = !session_media->locally_held ? STR_SENDRECV : STR_SENDONLY; media->attr[media->attr_count++] = attr; /* If we've got rtcp-mux enabled, add it unless we received an offer without it */ @@ -1471,18 +1472,18 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a ast_sockaddr_is_any(addrs) || pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL) || pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) { - if (!session_media->held) { + if (!session_media->remotely_held) { /* The remote side has put us on hold */ ast_queue_hold(session->channel, session->endpoint->mohsuggest); ast_rtp_instance_stop(session_media->rtp); ast_queue_frame(session->channel, &ast_null_frame); - session_media->held = 1; + session_media->remotely_held = 1; } - } else if (session_media->held) { + } else if (session_media->remotely_held) { /* The remote side has taken us off hold */ ast_queue_unhold(session->channel); ast_queue_frame(session->channel, &ast_null_frame); - session_media->held = 0; + session_media->remotely_held = 0; } else if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE) && (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) { ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER); @@ -1513,9 +1514,9 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a * instance itself. */ ast_rtp_instance_set_timeout(session_media->rtp, 0); - if (session->endpoint->media.rtp.timeout && !session_media->held) { + 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->held) { + } 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); }