diff options
| author | George Joseph <george.joseph@fairview5.com> | 2016-01-07 10:57:01 -0700 |
|---|---|---|
| committer | George Joseph <george.joseph@fairview5.com> | 2016-01-11 18:41:31 -0600 |
| commit | a41aab477ac317f9d973db253b4c30cd6a6db5b8 (patch) | |
| tree | 7921ac4515db887d05e00dd116cd5b90536d639d /res | |
| parent | 188438c53f5c5424cd45c3b904c58508881f0baa (diff) | |
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
Diffstat (limited to 'res')
| -rw-r--r-- | res/res_pjsip.c | 8 | ||||
| -rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 1 | ||||
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 9 |
3 files changed, 17 insertions, 1 deletions
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; } |