Merge "pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address" into 13

CHANGES

@@ -46,6 +46,14 @@ Queue
  * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
    the queue member was paused.
 
+res_pjsip_sdp_rtp
+------------------
+ * A new option (bind_rtp_to_media_address) has been added to endpoint which
+   will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
+   media_address as well as using it in the SDP.  If set, RTP packets will now
+   originate from the media address instead of the operating system's "primary"
+   ip address.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
 ------------------------------------------------------------------------------

configs/samples/pjsip.conf.sample

@@ -615,6 +615,9 @@
 ;disallow=      ; Media Codec s to disallow (default: "")
 ;dtmf_mode=rfc4733      ; DTMF mode (default: "rfc4733")
 ;media_address=         ; IP address used in SDP for media handling (default: "")
+;bind_rtp_to_media_address=     ; Bind the RTP session to the media_address.
+                                ; This causes all RTP packets to be sent from
+                                ; the specified address. (default: "no")
 ;force_rport=yes        ; Force use of return port (default: "yes")
 ;ice_support=no ; Enable the ICE mechanism to help traverse NAT (default: "no")
 ;identify_by=username   ; Way s for Endpoint to be identified (default:

contrib/ast-db-manage/config/versions/26d7f3bf0fa5_add_bind_rtp_to_media_address_to_pjsip.py

@@ -0,0 +1,31 @@
+"""add bind_rtp_to_media_address to pjsip
+
+Revision ID: 26d7f3bf0fa5
+Revises: 2d078ec071b7
+Create Date: 2016-01-07 12:23:42.894400
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '26d7f3bf0fa5'
+down_revision = '2d078ec071b7'
+
+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('bind_rtp_to_media_address', yesno_values))
+
+
+def downgrade():
+    op.drop_column('ps_endpoints', 'bind_rtp_to_media_address')

include/asterisk/res_pjsip.h

@@ -571,6 +571,8 @@ struct ast_sip_endpoint_media_configuration {
 	unsigned int cos_video;
 	/*! Is g.726 packed in a non standard way */
 	unsigned int g726_non_standard;
+	/*! Bind the RTP instance to the media_address */
+	unsigned int bind_rtp_to_media_address;
 };
 
 /*!

res/res_pjsip.c

@@ -233,6 +233,14 @@
 					</para></note>
 					</description>
 				</configOption>
+				<configOption name="bind_rtp_to_media_address">
+					<synopsis>Bind the RTP instance to the media_address</synopsis>
+					<description><para>
+						If media_address is specified, this option causes the RTP instance to be bound to the
+						specified ip address which causes the packets to be sent from that address.
+					</para>
+					</description>
+				</configOption>
 				<configOption name="force_rport" default="yes">
 					<synopsis>Force use of return port</synopsis>
 				</configOption>

res/res_pjsip/pjsip_configuration.c

@@ -1850,6 +1850,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "outbound_auth", "", outbound_auth_handler, outbound_auths_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "aors", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, aors));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "media_address", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.address));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "bind_rtp_to_media_address", "no", OPT_BOOL_T, 1, STRFLDSET(struct ast_sip_endpoint, media.bind_rtp_to_media_address));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "identify_by", "username", ident_handler, ident_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "direct_media", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.direct_media.enabled));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "direct_media_method", "invite", direct_media_method_handler, direct_media_method_to_str, NULL, 0, 0);

res/res_pjsip_sdp_rtp.c

@@ -175,8 +175,15 @@ static int rtp_check_timeout(const void *data)
 static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
 {
 	struct ast_rtp_engine_ice *ice;
+	struct ast_sockaddr temp_media_address;
+	struct ast_sockaddr *media_address =  ipv6 ? &address_ipv6 : &address_ipv4;
 
-	if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
+	if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
+		ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0);
+		media_address = &temp_media_address;
+	}
+
+	if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, media_address, NULL))) {
 		ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->media.rtp.engine);
 		return -1;
 	}