[SR-Users] Again problems with ACK and Bye

[Daniel-Constantin Mierla]

Hello,

On 3/16/12 4:43 PM, Stoyan Mihaylov wrote:
> In wireshark, there are 2 fields - Source and Destination. I am 
> curious if there are variables in Kamailio which can show me those values.

are you looking for $si (source ip) and $Ri (received ip)?
> System:
> MYIP - IP of kamailio
> sip.mydomain.com <http://sip.mydomain.com> - address of kamailio
>
> User (Imsdroid for Android) ->Kamailio->Asterisk->Kamailio (same) 
> ->Other user.
> Problem - register, voice, video, and ring are OK, but ACK and BYE are 
> going in dead loop - and this happens may be around in 50% of cases - 
> not in all cases.
> If I am using other client (like hardware SIP phones), then I have no 
> problems, but it is really hard to find some client for Android which 
> supports video with Kamailio.
> To solve problem, I made one function in Kamailio.cfg - ACKBYE, which 
> is called only within dialog. And within dialog is called in main route.
> route[ACKBYE] {
> #!ifdef WITH_MYFORWARD
> xlog("SCRIPT: AckBye $td;$si;$du;$dd\n");
> if((($td=="sip.mydomain.com 
> <http://sip.mydomain.com>")||($td=="MYIP"))&&($si=="MYIP")){
> $du=$sht(forw=>$ft);
> xlog("SCRIPT: AckBye2 $td;$si;$du\n");
> return;
> }
> #!endif
> return;
> }
> Of cource I have
> if(is_method("INVITE")){
> ds_select_dst("1","4");
> $sht(forw=>$ft)=$du;
> }
> This worked for long time. But then we updated client (libraries etc), 
> and problem appeared again - from time to time. Calls between 2 same 
> clients can pass 2 or 5 times, and then problem appear.
> Before - $td and $si showed destination and source but now there is 
> difference:
> ERROR: <script>: SCRIPT: AckBye 192.168.2.251;MYIP;<null>;<null>
> $td in this case show destination, which is correct, but in wireshark, 
> destination and source = MYIP.
>
>
> ACK sip:MYIP;lr=on;nat=yes SIP/2.0
> Via: SIP/2.0/UDP MYIP;branch=z9hG4bKcydzigwkX
> Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX
> ............................................
> Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX
> Via: SIP/2.0/UDP 192.168.2.105:44794;branch=z9hG4bK338581244;rport=44794
> From: <sip:11171 at sip.MYDOMAIN.com 
> <mailto:sip%3A11171 at sip.MYDOMAIN.com>>;tag=1687441663
> To: <sip:11175 at 192.168.2.251 
> <mailto:sip%3A11175 at 192.168.2.251>>;tag=as13c57096
> Contact: 
> <sip:11171 at MYIP:5060;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
> Call-ID: 4026cd88-aab9-19db-dfbe-1510baafdbb3
> CSeq: 473985105 ACK
> Content-Length: 0
> Max-Forwards:  0
> Proxy-Authorization: Digest 
> username="11171",realm="sip.stribogkonsult.com 
> <http://sip.stribogkonsult.com>",nonce="T2NZUU9jWCWxMUHSQweHXVQppHS8iFhd",uri="sip:11175 at MYIP:5060",response="c5a95942af1502c208fa05d34c405907",algorithm=MD5
> Accept-Contact: 
> *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
> P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
> Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, 
> UPDATE, REFER
> Privacy: none
> P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
> User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.0.453 (doubango r653)

I suspect the client messes the Record-Route/Route set -- the ACK R-URI 
shows a Route address, like there was a strict router. You have to 
capture all the messages for such call (from initial INVITE to ACK) and 
send them over. ngrep is a tool convenient for that, to get it directly 
text suitable for sending over email.

Cheers,
Daniel

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

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