diff --git a/CHANGES b/CHANGES
index 8d5f5b388e1d6b8edbff5e794705f743ade54a08..6885c512adf1989e6341ba8d4d5f96a2c7168b03 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 9302fb2619ed8c1c647377831139a15b5e8f2a42..c8d7cc90e9a451ddbeab12044d0f40690f3385bd 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 0000000000000000000000000000000000000000..e7c11da19259133cc38cd5b5eddca1d489078243
--- /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 df13af699454f3c9d3142404a662b4ea5423f933..0c82732833a343b98ee347336b1333f2de660975 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 bf26fca8c6d012cb7d294ba0c83cfd71032b090b..90be734e678ae27a76bc8cdaef1b98ab284d9e4d 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 72f896ad0c4d1bd2beae19ee3aaf20ec9a788417..926bf3793cb1e3754abed954a1180b814594739d 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 1f2f21d7330602ad41d0198330a73a4c825c23f8..2a1f56ed4956bdf6aa6e41668d39d3a780881e97 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;
}