Ciao Michele,
per uscire con il contact corretto devi usare l'advertised_address /
advertised_port se il serves ha solo un IP interno.
Questo fa nascere vari problemi perché la correzione viene fatta anche
per verso i client interni.
Non c'è il concetto di Lan come per asterisk, almeno finora.
Nelle ultime versioni ho visto dei parametri in più quindi le cose
potrebbero essere migliorate.
La miglior configurazione nel tuo caso sarebbe un server dual homed con
IP interno ed esterno.
Ciao
s
Il 03/11/2015 10:43, Michele Pinassi ha scritto:
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@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@VOIP01:5060>;tag=gujliebxxu//
//To: <sip:009123886472@VOIP01:5060;user=phone>//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//CSeq: 2 INVITE//
//Max-Forwards: 69//
//User-Agent: snom760/8.7.5.17//
//Contact: <sip:[email protected]: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@VOIP01:5060>;tag=gujliebxxu//
//To: <sip:009123886472@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@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@VOIP01:5060>;tag=gujliebxxu//
//To: <sip:009123886472@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@VOIP02 SIP/2.0//
//Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
//From: "Michele Pinassi" <sip:5002@VOIP01:5060>;tag=gujliebxxu//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//To: <sip:009123886472@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 [email protected]
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo,http://www.faq.unisi.it
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