[SR-Users] Problem with Route header

[Geoffrey Mina]

The Route header in this scenario is being used by a SIP client using kamailio as the proxy.  You can see that it takes the value of the Route header and uses it as the URI on the outgoing side.

What is causing it to modify the Route header and put it back in?  The first INVITE I supplied is coming into the kamailio server. The second is what t_relay sent upstream. 

Should I just not call the loose_route function and go straight to t_relay after calling remove_hf("Route")?

Thanks. 

Geoff Mina
Founder/CTO
Connect First Inc.
720.335.5924
888.410.3071
gmina at ConnectFirst.com

Sent from my iPhone

On May 2, 2012, at 12:42 AM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> Hello,
> 
> the Route header in initial invite is usually set by phones that have an outbound proxy setting. Kamailio doesn't add any Route header itself, unless append_hf()/insert_hf() is used -- record_route() adds Record-Route headers.
> 
> To deal with this case you should do loose_route() only for requests within dialog (those that have To header tag). For the rest just remove the Route header. If you look at default config file in v3.2.x, you will see this kind of processing (just to analyze it, not need to upgrade to 3.2.x).
> 
> Cheers,
> Daniel
> 
> On 5/2/12 12:26 AM, Geoffrey Mina wrote:
>> 
>> Greetings,
>> I am confused at some functionality I am seeing with Kamailio 1.5.4.  I know this is an old version, but I don't have the time to go through a lengthy upgrade process right now.  The issue I am seeing is that the server is inserting a Route header with it's own IP address for an unknown reason.  Here is the initial invite (removed SDP for simplicity):
>> 
>> INVITE sip:13 at 67.207.130.146:5060 SIP/2.0
>> Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport
>> From: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com>;tag=as1cad6370
>> To: <sip:13 at 67.207.130.146:5060>
>> Contact: <sip:9546496707 at 68.64.220.108>
>> Call-ID: 43134ece101abfca6ecab20212295909 at dev-asterisk.mydomain.com
>> CSeq: 102 INVITE
>> User-Agent: G-Tel v1.0
>> Max-Forwards: 70
>> Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com>;privacy=off;screen=no
>> Date: Tue, 01 May 2012 18:17:50 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>> Supported: replaces
>> Route: <sip:13 at boulder-voip.mydomain.com>
>> P-Account-ID: 99990023
>> P-Proxy-Route: Yes
>> Content-Type: application/sdp
>> Content-Length: 240
>> 
>> The basics of what happen next are:
>> 
>> t_check_trans();
>> record_route();
>> remove_hf("P-Proxy-Route");
>> if(loose_route()){
>>    route(3);
>> }
>> 
>> 
>> route[3]{
>>         t_on_reply("1");
>>         if(!t_relay()){
>>                 sl_reply_error();
>>         }
>> }
>> 
>> The INVITE that goes out has the funky Route: header with the Kamailio IP in there.  This is causing problems for some of the upstream proxy servers (obviously).  
>> 
>> INVITE sip:13 at boulder-voip.mydomain.com SIP/2.0
>> Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370>
>> Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0
>> Via: SIP/2.0/UDP 68.64.220.108:5060;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060
>> From: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com>;tag=as1cad6370
>> To: <sip:13 at 67.207.130.146:5060>
>> Contact: <sip:9546496707 at 68.64.220.108>
>> Call-ID: 43134ece101abfca6ecab20212295909 at dev-asterisk.mydomain.com
>> CSeq: 102 INVITE
>> User-Agent: G-Tel v1.0
>> Max-Forwards: 69
>> Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com>;privacy=off;screen=no
>> Date: Tue, 01 May 2012 18:17:50 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>> Supported: replaces
>> P-Account-ID: 99990023
>> Content-Type: application/sdp
>> Content-Length: 240
>> Route: <sip:13 at 67.207.130.146:5060>
>> 
>> 
>> Any idea what may be causing this to happen and how I could prevent it?  I have tried removing the Route header using the remove_hf("Route") before doing the t_relay, but that doesn't seem to help.
>> 
>> Thanks,
>> Geoff
>> 
>> 
>> 
>> 
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
> -- 
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20120502/19cb8a2d/attachment-0001.htm>