Sorry Bogdan but now my setup become a bit differente, i have the same
servers , Opensips+Asterisk in EC2 amazon (same LAN) and Cisco gateway
outside conected through public_ip to Opensips.
The SIP signalling works well but i have just oneway audio cause asterisk
send private ip on the reply to opensips invite (in same LAN) and opensips
forward that private ip to Cisco. So asterisk know the public ip of cisco
to establish rtp traffic but cisco don´t. ¿how can i solve this problem ?
¿there is anyway to change the rtp ip in the invite's reply ?
Best Regards!!
opensips.cfg:
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","looping");
exit;
}
if ($rU==NULL) {
sl_send_reply("484","Address Incomplete");
exit;
}
if (!has_totag()) {
record_route_preset(" Opensips public ip ");
xlog("route recorded \n");
} else {
loose_route();
t_relay();
exit;
}
if ( is_method("CANCEL") ) {
if ( t_check_trans() )
t_relay();
exit;
}
if (!is_method("INVITE")) {
send_reply("405","Method Not Allowed");
exit;
}
if (method=="INVITE") {
load_balance("1","calls");
}
if ($retcode<0) {
sl_send_reply("500","Service full");
exit;
}
xlog("Selected destination is: $du\n");
if (!t_relay()) {
sl_reply_error();
}
}
######################################################################################################
U 2010/12/03 13:00:27.034603 80.65.13.238:65071 -> 10.229.123.198:5060
INVITE sip:[email protected]:5060 SIP/2.0.
Date: Fri, 03 Dec 2010 12:04:32 GMT.
Call-Info: <sip:80.65.13.238:5060
>;method="NOTIFY;Event=telephone-event;Duration=2000".
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
Allow-Events: telephone-event.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE: 1800.
Remote-Party-ID: <sip:[email protected] <sip%[email protected]>
>;party=calling;screen=yes;privacy=off.
Cisco-Guid: 1378169425-4262203871-3197108258-2438471722.
Timestamp: 1291377872.
Content-Length: 269.
User-Agent: Cisco-SIPGateway/IOS-12.x.
To:
<sip:[email protected]<sip%[email protected]>
>.
Contact: <sip:[email protected]:5060>.
Expires: 180.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Call-ID: [email protected].
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC71C9A.
CSeq: 101 INVITE.
Max-Forwards: 70.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 849 9795 IN IP4 80.65.13.238.
s=SIP Call.
c=IN IP4 80.65.13.238.
t=0 0.
m=audio 23660 RTP/AVP 18 101.
c=IN IP4 80.65.13.238.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
U 2010/12/03 13:00:27.035190 10.229.123.198:5060 -> 80.65.13.238:5060
SIP/2.0 100 Giving a try.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
To:
<sip:[email protected]<sip%[email protected]>
>.
Call-ID: [email protected].
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC71C9A.
CSeq: 101 INVITE.
Server: OpenSIPS (1.6.3-notls (i386/linux)).
Content-Length: 0.
.
U 2010/12/03 13:00:27.035263 10.229.123.198:5060 -> 10.228.26.150:5060
INVITE sip:[email protected]:5060 SIP/2.0.
Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
Date: Fri, 03 Dec 2010 12:04:32 GMT.
Call-Info: <sip:80.65.13.238:5060
>;method="NOTIFY;Event=telephone-event;Duration=2000".
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
Allow-Events: telephone-event.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE: 1800.
Remote-Party-ID: <sip:[email protected] <sip%[email protected]>
>;party=calling;screen=yes;privacy=off.
Cisco-Guid: 1378169425-4262203871-3197108258-2438471722.
Timestamp: 1291377872.
Content-Length: 269.
User-Agent: Cisco-SIPGateway/IOS-12.x.
To:
<sip:[email protected]<sip%[email protected]>
>.
Contact: <sip:[email protected]:5060>.
Expires: 180.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Call-ID: [email protected].
Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.0.
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC71C9A.
CSeq: 101 INVITE.
Max-Forwards: 69.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 849 9795 IN IP4 80.65.13.238.
s=SIP Call.
c=IN IP4 80.65.13.238.
t=0 0.
m=audio 23660 RTP/AVP 18 101.
c=IN IP4 80.65.13.238.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
U 2010/12/03 13:00:27.036250 10.228.26.150:5060 -> 10.229.123.198:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
46.51.135.212;branch=z9hG4bK1a6f.13422624.0;received=10.229.123.198.
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC71C9A.
Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
To:
<sip:[email protected]<sip%[email protected]>
>.
Call-ID: [email protected].
CSeq: 101 INVITE.
Server: Asterisk PBX 1.6.2.13.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Require: timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:[email protected] <sip%[email protected]>>.
Content-Length: 0.
.
U 2010/12/03 13:00:27.235884 10.228.26.150:5060 -> 10.229.123.198:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
46.51.135.212;branch=z9hG4bK1a6f.13422624.0;received=10.229.123.198.
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC71C9A.
Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
To:
<sip:[email protected]<sip%[email protected]>
>;tag=as33981ab2.
Call-ID: [email protected].
CSeq: 101 INVITE.
Server: Asterisk PBX 1.6.2.13.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Require: timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:[email protected] <sip%[email protected]>>.
Content-Type: application/sdp.
Content-Length: 262.
.
v=0.
o=root 1270939673 1270939673 IN IP4 10.228.26.150.
s=Asterisk PBX 1.6.2.13.
c=IN IP4 10.228.26.150.
t=0 0.
m=audio 10532 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
U 2010/12/03 13:00:27.236908 10.229.123.198:5060 -> 80.65.13.238:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC71C9A.
Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
To:
<sip:[email protected]<sip%[email protected]>
>;tag=as33981ab2.
Call-ID: [email protected].
CSeq: 101 INVITE.
Server: Asterisk PBX 1.6.2.13.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Require: timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:[email protected] <sip%[email protected]>>.
Content-Type: application/sdp.
Content-Length: 262.
.
v=0.
o=root 1270939673 1270939673 IN IP4 10.228.26.150.
s=Asterisk PBX 1.6.2.13.
c=IN IP4 10.228.26.150.
t=0 0.
m=audio 10532 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
U 2010/12/03 13:00:27.294728 80.65.13.238:65071 -> 10.229.123.198:5060
ACK sip:[email protected]:5060 SIP/2.0.
Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
Date: Fri, 03 Dec 2010 12:04:32 GMT.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
Allow-Events: telephone-event.
Content-Length: 0.
To:
<sip:[email protected]<sip%[email protected]>
>;tag=as33981ab2.
Call-ID: [email protected].
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC8B3D.
CSeq: 101 ACK.
Max-Forwards: 70.
.
U 2010/12/03 13:00:27.295705 10.229.123.198:5060 -> 10.228.26.150:5060
ACK sip:[email protected]:5060 SIP/2.0.
Date: Fri, 03 Dec 2010 12:04:32 GMT.
From: <sip:[email protected] <sip%[email protected]>
>;tag=274FBBA0-208D.
Allow-Events: telephone-event.
Content-Length: 0.
To:
<sip:[email protected]<sip%[email protected]>
>;tag=as33981ab2.
Call-ID: [email protected].
Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.2.
Via: SIP/2.0/UDP
80.65.13.238:5060;x-route-tag="cid:[email protected]<cid%[email protected]>
";branch=z9hG4bK1EDAC8B3D.
CSeq: 101 ACK.
Max-Forwards: 69.
2010/12/2 Bogdan-Andrei Iancu <[email protected]>
> Hi Nawfel,
>
> The problem is in one of the end points as for a 200 OK calls, the 200
> reply and the ACK is end-2-end.
>
> If you have a trace, maybe I can help you to see if there is a signalling
> problem.
>
> Regards,
> Bogdan
>
>
> Nawfel Oujdi wrote:
>
>> Hello!!
>> I m new in opensips and i m testing the load balancer cause i need it to
>> balance calls between 4 asterisk.For the start i make the following
>> scenario
>> Cisco gateway inbound ------> opensips ------> asterisk --------->
>> Cisco gateway outbound
>> when the call comes to the opensips, the load_balancer forward the call
>> correctly to my asterisk but the call hangs up after 15 seg
>> approximately.When i did a ngrep for the sip traffic in opensips, i
>> realized that cisco gateway inbound never sent the ACK for 200 OK to
>> opensips .
>> In the Cisco's logs i saw that the reply of 200 ok is sent directly to
>> public ip of asterisk but never to opensips server so asterisk still waiting
>> for the ACK from opensips.
>> In the same way opensips never receive the BYE packet and the load never
>> decrease when the call is hanging up.
>>
>> Cisco gateway opensips asterisk
>> ---invite---> <--trying----
>> ---invite---> <---trying---
>> <----200OK---
>> <---200 OK---
>> <----200OK---
>> <---200 OK---
>> <----200OK---
>> <---200 OK---
>> <----200OK---
>> <---200 OK--- Please
>> can somebady help me to understand what cause that?
>>
>> Best Regards!!
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> [email protected]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Nawfel Oujdi
*Ingeniero VoIP*
[email protected]
EG telecom S.A | www.egtelecom.es
Oficina: *902 050 080*
Agustín de Foxá, 25 - 9B | 28036 Madrid
------------------------------
Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s)
dirección(es) indicadas anteriormente; el carácter confidencial, personal e
intransferible del mismo está protegido legalmente. Cualquier revelación,
uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha
recibido este mensaje por equivocación, notifíquelo inmediatamente a la
persona que lo ha enviado y borre el mensaje original junto con sus ficheros
anexos sin leerlo ni grabarlo, total o parcialmente.
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
