Re: [OpenSIPS-Users] Opensips as proxy, config problem on INVITES

[Benjamin Cropley]

There's quite a lot of trace to go through there, but this screams that
you're looping to me.. I suggest you look at the routing on the OTHER
boxes, see if you can spot something that might be doing that.

On Tue, May 31, 2016 at 11:07 AM, Saioa Perurena <
saioa.perur...@enigmedia.es> wrote:

> Hi,
>
> We have only one proxy Opensips (version 1.11) that does all the work
> (register's, invite's, tls conection...) on a DMZ behind a firewall.
>
> We want to move that to a new schema, with one Opensips (version 2.1)
> as a frontend that handles the tls connections (with a public ip) on
> the DMZ, and another Opensips (version 1.11) at the backend (with
> private ip), but we are not able to complete the invite signaling
> correctly. Register, message, options worked ok, but we have problems
> with the invite.
> The invite request arrives to the callee, and the OK answers arrives
> to the caller, but the caller does not send and ACK to this OK, so the
> callee keeps sending OK until it send a BYE because of timeout.
>
> Any idea of where is the problem or what am i doing wrong?? Any advice
> will be appreciated!!
>
> I attach the sip_trace log and the Opensips script of the frontend server:
>
> Caller: sip:u1iupzg...@jipubnx2ef.bell.enigmedia.eu
> Callee: d27p6ui...@jipubnx2ef.bell.enigmedia.eu
> Opensips frontend ip: internal -> 192.168.3.35, external -> 192.168.1.18
> Opensips backend ip: internal -> 192.168.2.6
>
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
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[OpenSIPS-Users] Opensips as proxy, config problem on INVITES

[Saioa Perurena]

Hi,

We have only one proxy Opensips (version 1.11) that does all the work
(register's, invite's, tls conection...) on a DMZ behind a firewall.

We want to move that to a new schema, with one Opensips (version 2.1)
as a frontend that handles the tls connections (with a public ip) on
the DMZ, and another Opensips (version 1.11) at the backend (with
private ip), but we are not able to complete the invite signaling
correctly. Register, message, options worked ok, but we have problems
with the invite.
The invite request arrives to the callee, and the OK answers arrives
to the caller, but the caller does not send and ACK to this OK, so the
callee keeps sending OK until it send a BYE because of timeout.

Any idea of where is the problem or what am i doing wrong?? Any advice
will be appreciated!!

I attach the sip_trace log and the Opensips script of the frontend server:

Caller: sip:u1iupzg...@jipubnx2ef.bell.enigmedia.eu
Callee: d27p6ui...@jipubnx2ef.bell.enigmedia.eu
Opensips frontend ip: internal -> 192.168.3.35, external -> 192.168.1.18
Opensips backend ip: internal -> 192.168.2.6





from_ip | from_port | to_ip  | to_port | direction |

content 
   
+---++-+---+---
 192.168.10.109 | 41302 | 192.168.3.35   |5061 | in| INVITE 
sip:d27p6ui...@jipubnx2ef.bell.enigmedia.eu SIP/2.0\r 
|   || |   | Via: 
SIP/2.0/TLS 192.168.200.238:41302;branch=z9hG4bK.5v5fmStat;rport\r 
|   || |   | From: 
;tag=~DvmtD4uF\r 
|   || |   | To: 
sip:d27p6ui...@jipubnx2ef.bell.enigmedia.eu\r
|   || |   | CSeq: 20 
INVITE\r
|   || |   | Call-ID: 
At8pM4Jbgs\r
|   || |   | 
Max-Forwards: 69\r   
|   || |   | Route: 
\r  
|   || |   | Supported: 
outbound\r
|   || |   | Allow: 
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, 
UPDATE\r
|   || |   | 
CALLEE-KEY: 123456789123456789\r 
|   || |   | 
Content-Type: application/sdp\r  
|   || |   | 
Content-Length: 538\r
|   || |   | 
LocalDateTime: 2016-05-31T11:14:47 +0200\r   
|   || |   | InstallId: 
0oitm5of4m\r  
|   || |   | Contact: 
;+sip.instance=""\r
|   || |   | 
User-Agent: 0oitm5of4m/EnigmediaAppAnd/2.3.1/bell-sip/1.4.2/3.9.1.E1.0.11\r 
   
|   || |   | \r 
  
|   || |   | v=0\r  
  
|   || |   | 
o=u1iupzg6we 944 1255 IN IP4 192.168.200.238\r   
|   || |   | s=Talk\r   
  
|   || |   | c=IN IP4 
192.168.200.238\r   
|   || |   | t=0 0\r
  
|   || |   | 
a=rtcp-xr:rcvr-rtt=all:1 stat-summary=loss,dup,jitt,TTL voip-metrics\r  
   
|   || |   | m=audio