Merge "chan_pjsip: Add support for passing hold and unhold requests through." into 13

channels/chan_pjsip.c

@@ -1359,6 +1359,39 @@ 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)
 {
@@ -1503,7 +1536,15 @@ 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));
@@ -1511,7 +1552,15 @@ 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;

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;

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

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')

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 */

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 */

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>

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);

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);
 	}