[OpenSIPS-Users] B2B + top hiding + origination media IP

[Ovidiu Sas]

The B2B module is operating on the received INVITE.  Any changes that
you make to the received INVITE are not visible by the B2B module.
Use a proxy to perform whatever you want to do (rtpproxy, accounting,
etc.) and a separate server only for b2b (top hiding).

Regards,
Ovidiu Sas

On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov <kamen.petrov at gmail.com> wrote:
> Hi Guys,
>
> I am testing the following call flow:
> Soft Phone => opensips (configured for B2B) => third party termination SIP
> proxy
>
> Here is my config:
>
> modparam("b2b_entities", "script_req_route", "b2b_request")
> modparam("b2b_entities", "script_reply_route", "b2b_reply")
>
>
>
> local_route {
>         xlog("================LOCAL_ROUTE ($rm - $rr)============\n");
>         setflag(22);
>         if (is_method("INVITE")) {
>                 engage_rtp_proxy("e","<OPENSIPS_IP>");
>                 exit;
>         }
>         else if (is_method("BYE") ) {
>                 xlog("================BYE============\n");
>         }
> }
>
>
> route[b2b_request] {
>         $avp(s:source_ip_address) := $si;
>         perl_exec("messagedump_route", "messages");
>         xlog("b2b_request ($ci) ($rm - $rr)\n");
> }
>
>
> route[b2b_reply] {
>         $avp(s:source_ip_address) := $si;
>         perl_exec("messagedump_reply", "messages");
>         xlog("b2b_reply ($ci) - $rm - $rr\n");
> }
>
>
> route{
> ...
>         if (is_method("INVITE") && perl_exec("check_for_forwarding_number"))
> {
>                 engage_rtp_proxy("e","<OPENSIPS_IP>");
>                 xlog("LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD TO:
> $avp(s:fwd_ip)\n");
>                 setflag(1); # do accounting
>                 xlog("L_ERR", "LOG: to uri=[$tu] [$avp(s:sip_proxy_ip)]\n");
>
>                 b2b_init_request("top hiding");
>                 exit;
>         };
>
> ...
> }
>
>
> What happens is:
> - INVITE from the soft phone to the opensips
> - catched by the B2B and relayed to the third party SIP proxy + trying
> returned to the soft phone
> - "Session Progress" received from the third party SIP proxy -> opensips ->
> my soft phone
>
> At that stage, here is what I have on the soft phone log:
> 18:56:50 UDP Packet Received from <OPENSIPS_IP>:5060
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604
> To: <sip:359883409291@<OPENSIPS_DOMAIN>:5060>;tag=B2B.113.667
> From: "359883327749" <sip:359883327749@<OPENSIPS_DOMAIN>:5060>;tag=1040
> Call-ID: 1296636915-3604-SALASWORK at 192.168.1.2
> CSeq: 361 INVITE
> Content-Type: application/sdp
> Contact: <sip:<OPENSIPS_IP>:5060;transport=udp>
> Server: OpenSIPS (1.6.3-notls (x86_64/linux))
> Content-Length: 184
>
> v=0
> o=SBCSIPUAS 900116523 1 IN IP4 <THIRD_PARTY_SIP_PROXY_IP>
> s=SBCSIPUAS SIP STACK v1.0
> c=IN IP4 <THIRD_PARTY_SIP_PROXY_IP>
> t=0 0
> m=audio 17900 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> a=sendrecv
> a=maxptime:30
>
>
> As can be seen, the media IP is not rewritten by the opensips and the IP
> passed to my soft phone is the IP of the termination IP for the opensips
> (i.e. the third party SIP proxy IP). Because of that, my soft phone starts
> the RTP directly to my provider instead trough the RTP proxy that is
> attached to the opensips.
> Just to clarify, the media IP of my soft phone is not passed to my provider
> - that case is handled good.
>
>
> Any idea what is missing ?
>
> Thanks in advance.
> -- Kamen
>
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