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 b0d5fda5d9287994a071f6b98e3d7bbe21510344..2a243b6d1079b7b49898b9b0e4b65ea88edc3a74 100644 (file)
--- 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 76f351f1e897fba2263c88e7b8552ac474fe8b0f..6cdc7213b3bc67e1a9c7ef1b62b7a87d4f9e882b 100644 (file)
--- 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 bbd0317e798db80606ced91f6ed7f4aa9c6ba371..9e6365e2532bdfe62a97febfd2d8ec1b5ff7c8f5 100644 (file)
--- 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 (file)
index 0000000..cf7baf6
--- /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 5e8dd3a24f653199832ee4cb5f5959dfddd80d83..163059531c1aaf986b4d89a72d1da674a5b4e272 100644 (file)
--- 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 e2d4fcd27f6ca73a9e420646cb05107e3c940ed5..51d75c6d6dc94c78e14e31fd1e36c470dc346e28 100644 (file)
--- 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 c41b545b498d97852eaea4a59be039b76cdf8778..ec159d6fa5e64b6dc9ca3133a9b7454de875ca0c 100644 (file)
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -744,6 +744,9 @@
                                <configOption name="user_eq_phone" default="no">
                                        <synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
                                </configOption>
+                               <configOption name="moh_passthrough" default="no">
+                                       <synopsis>Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side</synopsis>
+                               </configOption>
                                <configOption name="sdp_owner" default="-">
                                        <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
                                </configOption>
@@ -2286,6 +2289,9 @@
                                <parameter name="UserEqPhone">
                                        <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para>
                                </parameter>
+                               <parameter name="MohPassthrough">
+                                       <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_passthrough']/synopsis/node())"/></para>
+                               </parameter>
                                <parameter name="SdpOwner">
                                        <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para>
                                </parameter>

diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index b5525beeb2e75f074f515f157cf01b1a7e9db1b0..db53592588427b99b1ca8bd81d319450a85f73bb 100644 (file)
--- 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 76d47a3f6d7c42b526c5d0705ae80d9cd99d5f87..98e981965f697bace44372be4fa7ac0956770bad 100644 (file)
--- 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);
        }