pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address

On a system with multiple ip addresses in the same subnet, if a
transport is bound to a specific ip address and endpoint/media_address
 is set, the SIP/SDP will have the correct address in all fields but
the rtp stream MAY still originate from one of the other ip addresses,
most probably the "primary" ip address.  This happens because
 res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
the "all" ip address (0.0.0.0 or ::).

The new option causes res_pjsip_sdp_rtp/create_rtp to call
ast_rtp_instance_new with the endpoint's media_address (if specified)
instead of the "all" address.  This causes the packets to originate from
the specified address.

ASTERISK-25632
ASTERISK-25637
Reported-by: Olivier Krief
Reported-by: Dan Journo

Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88

7 files changed, 61 insertions, 1 deletions

diff --git a/CHANGES b/CHANGES
index 8d5f5b388..6885c512a 100644
--- a/CHANGES
+++ b/CHANGES
@@ -234,6 +234,14 @@ Voicemail
    app_voicemail will be skipped.  Use 'preload=app_voicemail.so' in
    modules.conf to force app_voicemail to be the voicemail provider.
 
+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 ------------
 ------------------------------------------------------------------------------
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 9302fb261..c8d7cc90e 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/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:
diff --git a/contrib/ast-db-manage/config/versions/26d7f3bf0fa5_add_bind_rtp_to_media_address_to_pjsip.py b/contrib/ast-db-manage/config/versions/26d7f3bf0fa5_add_bind_rtp_to_media_address_to_pjsip.py
new file mode 100644
index 000000000..e7c11da19
--- /dev/null
+++ b/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')
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index d9123f983..6f4ea9a75 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -575,6 +575,8 @@ 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;
 };
 
 /*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index a4748d20e..c802c7776 100644
--- a/res/res_pjsip.c
+++ b/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>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 72f896ad0..926bf3793 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1847,6 +1847,7 @@ int ast_res_pjsip_initialize_configuration(void)
 	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);
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 1f2f21d73..2a1f56ed4 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/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;
 	}