[SR-Users] Misrouting ACK packet for 200 Ok messages.

[Daniel-Constantin Mierla]

It would be relevant to see the 200ok as received by each hop in the
call path. Also, be sure you don't use fix_nated_contact() on the proxy
if it is not the first node next to endpoint -- anyhow it is recommended
to use set_contact_alias().

As a clafication, do you use tcp/tls between Kamailio2 and Asterisk?

Cheers,
Daniel

On 12/04/16 22:12, Yasin CANER wrote:
> Hello;
>     before sending this email i searched on google and doesnt solve
> this issue. all call flows are correct but one call that this isnt
> working right. it sends to _wrong port_ to ACK for 200 OK. i tried to
> fix contact header or remove contact header but it wasnt work. i
> looked at ietf for ACK and couldnt figure out why it happens.
> Does it need add a record route or remove Contact Header for every ack ?
>
>
> Thanks for help.
>
> i figure out  Kamailio-2 adds a Route header to ACK packet for sending
> Kamailio-1:5060, even if it doesnt add any command for it  in cfg.
>
> Here is call flow
>
> Asterisk and Kamailio is on the same ip and machine and public ip.
> different are ports. Kamailio-1 is another machine
>
>
> INVITE : UAC1-----> Kamailio-1:5060----->Kamailio-2:5061--->Asterisk:5060
>
> 200 OK:  UAC1<----- Kamailio-1:5060<-----Kamailio-2:5061<---Asterisk:5060
>
> ACK    : UAC1-----> Kamailio-1:5060----->Kamailio-2:5061--->Asterisk:10432
>
> 200 OK:  UAC1<----- Kamailio-1:5060<-----Kamailio-2:5061<---Asterisk:5060
>
> ACK    : UAC1-----> Kamailio-1:5060----->Kamailio-2:5061--->Asterisk:10432
>
> Retransmission.......
>
>
> Here is ACK packet, is it about_port on RU?_
>
>
> Asteriskip:5060----------->Kamailio2-ip:5061
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> kamailio2-ip:5061;branch=z9hG4bK4d2f.837f45483db59574c4dd70f70f8d0099.0;received=kamailio2-ip;rport=5061
> Via: SIP/2.0/UDP
> kamailio1-ip-main;branch=z9hG4bK4d2f.fb781e8caa5f96426c82530a23c0cc97.0
> Via: SIP/2.0/UDP
> uacip:5060;received=uacip;branch=z9hG4bK143c7384;rport=10432
> Record-Route:
> <sip:kamailio2-ip:5061;lr;ftag=as1c529e28;did=304.35c1;vsf=AAAAAAoBAQMLBQ4GAQN2A3kDFgQYABYeGRoyMDM->
> Record-Route: <sip:kamailio1-ip-main;lr;ftag=as1c529e28>
> From: 903122977162 <sip:903122977162 at kamailio2-ip>;tag=as1c529e28
> To: <sip:03129110911 at tstxyz.netgsm.com.tr>;tag=as39358508
> Call-ID: 169f342556f4e956445dfe0e2ee8ecf2 at uacip:5060
> CSeq: 103 INVITE
> Server: sipgw2
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:10213129110911 at kamailio2-ip:5060>
> Content-Type: application/sdp
> Require: timer
> Content-Length: 305
>
> v=0
> o=root 455426546 455426546 IN IP4 kamailio2-ip
> s=Asterisk PBX 11.21.2
> c=IN IP4 kamailio2-ip
> t=0 0
> m=audio 15926 RTP/AVP 18 8 0 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> Kamailio2-ip:5061----> Asterisk:10432
> ACK sip:10213129110911 at kamailio2-ip:10432 SIP/2.0
> Via: SIP/2.0/UDP
> kamailio2-ip:5061;branch=z9hG4bK4d2f.deb7bd96617c7067212dbc1673a216d2.0
> Via: SIP/2.0/UDP
> kamailio1-ip-main;branch=z9hG4bK4d2f.acd6ee83df5babb9e8f53268a4b1b948.0
> Via: SIP/2.0/UDP
> uacip:5060;received=uacip;branch=z9hG4bK59e2e846;rport=10432
> Max-Forwards: 68
> From: <sip:903122977162 at kamailio2-ip>;tag=as1c529e28
> To: <sip:03129110911 at tstxyz.netgsm.com.tr>;tag=as39358508
> Call-ID: 169f342556f4e956445dfe0e2ee8ecf2 at uacip:5060
> CSeq: 103 ACK
> User-Agent: Asterisk PBX 11.21.2
> Content-Length: 0
>
>
>
> ietf :
>
> If the INVITE request whose response is being acknowledged had Route
>    header fields, those header fields MUST appear in the ACK.  This is
>    to ensure that the ACK can be routed properly through any downstream
>    stateless proxies.
>
>
> _______________________________________________
> 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
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160413/e9a13efe/attachment.html>