Hello,

On 3/29/12 11:17 AM, Hervé Cochet wrote:

Hello,

I used kamailio to handle SIP line from a provider using uacreg table.
Everything was working fine till they add a proxy which do not set (or not propagate) route header in ACK or BYE request.

So my routing logic is broken in the "in-dialog" transaction mode, since the loose_route return FALSE.

If I try to force a t_relay after the loose_route, my message may not be routing properly if for example my UAC was not connected with the standard 5060 port or use a different protocol like TCP (It works if both are in UDP and use 5060 port).
Is there a way to retrieve the record-route previously sets for this session in order to route this message properly ?

is the Record-Route set mirrored in the 200ok?

If the device is not supporting record-routing, then ACK/BYE should not get to your proxy. Can you post a full sip trace for such a call (from invite to bye)? It will help to understand how the message flow is and maybe we can help more.

As a generic hint for storing/retrieving data, look at htable module, or if you prefer to use database storage, sqlops may be an alternative. You can keep the values based on callid.

Another problem is that the dialog module do not match the BYE transaction, the "did" variable is missing since the route is not there but it should match the request with SIP matching (dlg_match_mode is set to 1).

For example my dialog parameters are:

dialog::  hash=1214:1340198900
    state:: 3
    timestart:: 1333006717
    timeout:: 80537132
    callid:: 16260-CI-30dd2867-797784ae0@my.sip.provider
    from_uri:: sip:0367023024@my.sip.provider;user=phone
    from_tag:: 16260-GI-30dd2868-17bb342c3
    caller_contact:: sip:13.12.14.20:5060
    caller_cseq:: 815264875
    caller_route_set:: <sip:13.12.14.20:5060;lr>
    caller_bind_addr:: udp:130.120.140.131:5060
    to_uri:: sip:0974711672@13.12.14.17;user=phone
    to_tag:: 1878467993
    callee_contact:: sip:0974711672@96.57.249.78:1024
    callee_cseq:: 815264875
    callee_route_set::
    callee_bind_addr:: udp:130.120.140.131:5060

The BYE request:
BYE sip:0974711672@96.57.249.78:1024 SIP/2.0
Call-ID: 16260-CI-30dd2867-797784ae0@my.sip.provider
CSeq: 815264876 BYE
From: "0033482531303" <sip:0367023024@my.sip.provider;user=phone>;tag=16260-GI-30dd2868-17bb342c3
Max-Forwards: 28
Record-Route: <sip:13.12.14.20:5060;lr>
To: <sip:0974711672@13.12.14.17;user=phone>;tag=1878467993
Via: SIP/2.0/UDP 13.12.14.20:5060;branch=z9hG4bK-LNVP-1196a924-5f1f495a
Reason: q.850;cause=16
User-Agent: Cirpack/v4.42a (gw_sip)
Content-Length: 0

Why does this BYE request is not matched by the dialog modules using SIP parameters (Call-ID, uri, tag seems correct) ?

Can you try to execute dlg_manage()? If Route header is missing, then there is nothing that triggers automatically dialog matching.

Cheers,
Daniel

Hervé

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
    

-- 
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
http://www.asipto.com/index.php/kamailio-advanced-training/