[OpenSIPS-Users] Try to setup ENUM NRENUM support...

[Michele Pinassi]

Hi all,

i'm trying to setup enum (NRENUM) infrastructure for our university. We
have a public IP voip server, VOIP01, with a private network for voip
phones (172.20.x.x). To let RTP flow through the private and public
network, i set up a RTP Proxy:

/onreply_route[enum_answer] {//
//    if(has_body("application/sdp")) {//
//        rtpproxy_answer();//
//    }//
//}//
//
//onreply_route[enum_offer] {//
//    if(has_body("application/sdp")) {//
//        rtpproxy_offer();//
//    }//
//}//
//
//route[enum] {//
//    xlog("L_INFO","Route to ENUM [$fd/$fu/$rd/$ru/$si/]\n");//
//   //
//    if (is_method("INVITE")) {//
//    if(has_body("application/sdp")) {//
//            if (rtpproxy_offer()) {//
//                t_on_reply("enum_answer");//
//            }//
//        } else {//
//            t_on_reply("enum_offer");//
//        }//
//    }//
//    if (is_method("ACK") && has_body("application/sdp")) {//
//        rtpproxy_answer();//
//    }//
//    //
//    t_on_failure("pstn");//
//
//    if(!t_relay()) {//
//    sl_reply_error();//
//    }//
//    exit;//
//}/


but on establishing call, this is the tcpdump trace between VOIP01 and
VOIP02 i get this:

/IP VOIP01.5060 > VOIP02.5060: UDP, length 1170//
//
//INVITE sip:86472 at VOIP02 SIP/2.0//
//Record-Route: <sip:VOIP01;lr;ftag=gujliebxxu;did=a26.6c5c0e13>//
//Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
//Via: SIP/2.0/UDP
172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//
//From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
//To: <sip:009123886472 at VOIP01:5060;user=phone>//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//CSeq: 2 INVITE//
//Max-Forwards: 69//
//User-Agent: snom760/8.7.5.17//
//Contact: <sip:5002 at 172.20.1.47:57907>;reg-id=1//
//X-Serialnumber: 00041371928A//
//P-Key-Flags: resolution="31x13", keys="4"//
//Accept: application/sdp//
//Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE//
//Allow-Events: talk, hold, refer, call-info//
//Supported: timer, 100rel, replaces, from-change//
//Session-Expires: 3600//
//Min-SE: 90//
//Content-Type: application/sdp//
//Content-Length: 228//
//
//v=0//
//o=root 846474428 846474428 IN IP4 172.20.1.47//
//s=call//
//c=IN IP4 VOIP01//
//t=0 0//
//m=audio 63194 RTP/AVP 9 0 8//
//a=rtpmap:9 G722/8000//
//a=rtpmap:0 PCMU/8000//
//a=rtpmap:8 PCMA/8000//
//a=ptime:20//
//a=sendrecv//
//a=nortpproxy:yes//
//
//IP VOIP02.5060 > VOIP01.5060: UDP, length 722//
//
//SIP/2.0 100 Trying//
//Via: SIP/2.0/UDP
VOIP01:5060;branch=z9hG4bKe567.edebdee3.0;received=VOIP01;rport=5060//
//Via: SIP/2.0/UDP
172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//
//Record-Route: <sip:VOIP01;lr;ftag=gujliebxxu;did=a26.6c5c0e13>//
//From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
//To: <sip:009123886472 at VOIP01:5060;user=phone>//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//CSeq: 2 INVITE//
//Server: FPBX-2.11.0(11.7.0)//
//Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH//
//Supported: replaces, timer//
//Session-Expires: 1800;refresher=uas//
//Contact: <sip:86472 at VOIP02:5060>//
//Content-Length: 0//
//
//
//IP VOIP02.5060 > VOIP01.5060: UDP, length 590//
//
//SIP/2.0 603 Declined//
//Via: SIP/2.0/UDP
VOIP01:5060;branch=z9hG4bKe567.edebdee3.0;received=VOIP01;rport=5060//
//Via: SIP/2.0/UDP
172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//
//From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
//To: <sip:009123886472 at VOIP01:5060;user=phone>;tag=as15312d47//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//CSeq: 2 INVITE//
//Server: FPBX-2.11.0(11.7.0)//
//Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH//
//Supported: replaces, timer//
//Content-Length: 0//
//
//
//IP VOIP01.5060 > VOIP02.5060: UDP, length 375//
//
//ACK sip:86472 at VOIP02 SIP/2.0//
//Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
//From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//To: <sip:009123886472 at VOIP01:5060;user=phone>;tag=as15312d47//
//CSeq: 2 ACK//
//Max-Forwards: 70//
//User-Agent: VoIP Unisi.it//
//Content-Length: 0/

The main doubt is: /a=nortpproxy:yes/ ...why ?

Thanks, Michele



-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - centralino at unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20151103/661502bd/attachment.htm>