[SR-Users] CANCEL not matching INVITES !

Daniel-Constantin Mierla miconda at gmail.com
Wed Nov 30 13:47:12 CET 2011


Hello,

is the SIP trace complete?

What I could find inside is:
- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk asks for authentication - 401
- ack
- there is no new INVITE with credentials for kamailio and asterisk
- but the phone starts sending CANCELs -- since there is no active 
INVITE transaction, kamailio just drops it due to config rules
- after a while asterisk starts sending like 180 ringing, then 200ok ... 
really strange

Maybe you haven't captured all the sip traffic. If you want to use 
ngrep, do on kamailio server:

ngrep -d any -qt -W byline port 5060

If that's all the traffic, then xlite and asterisk seems to have some 
bugs - both were aware of 401 reply (asterisk generated it, xlite sent 
the ACK for it) -- so no ongoing call to CANCEL by xlite, or to answer 
by Asterisk (the 180, 200 replies).

 From kamailio point of view, if there is no INVITE following the 401 
reply to xlite, there is no active invite transaction to cancel.

Cheers,
Daniel

On 11/30/11 12:02 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> I will look over it soon - since you sent pcap I couldn't look at it 
> directly from the email. ngrep outputs plain text which is easy to 
> read from email, the reason I am asking mainly for ngrep traces since 
> many times I am not around a computer where is convenient to open pcap 
> file. On the other hand, if it is a transmission problem (at transport 
> layer), pcap file is better.
>
> Cheers,
> Daniel
>
> On 11/29/11 5:07 AM, Sammy Govind wrote:
>> Hello again,
>>
>> Please see the attached wireshark trace, I tried for a sipgrep trace 
>> but couldn't somehow. I hope this will get me some clue on what I'm 
>> doing wrong.
>>
>> This is a setup with Kamailio in front of Asterisk Servers. Kamailio 
>> is multihomed and MS are on private IPs, all the calls are routed to 
>> MSs and then comeback for further dial-outs.
>>
>> Please see the Continuous CANCEL requests which aren't terminating 
>> the call.
>>
>> Thanks,
>> Sammy.
>>
>> On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind <govoiper at gmail.com 
>> <mailto:govoiper at gmail.com>> wrote:
>>
>>     Thanks for your reply I will attach the wireshark traces as soon
>>     as I get to my workstation.
>>
>>     BR,
>>     Sammy.
>>
>>
>>     On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         send the ngrep trace of such call, from the initial INVITE,
>>         you can use:
>>
>>         ngrep -d any -qt -W byline port 5060
>>
>>         The sip trace will help to see what is wrong with that CANCEL.
>>
>>         Cheers,
>>         Daniel
>>
>>
>>         On 11/28/11 7:19 AM, Sammy Govind wrote:
>>>         Anyone please help.
>>>
>>>         On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind
>>>         <govoiper at gmail.com <mailto:govoiper at gmail.com>> wrote:
>>>
>>>             Hello list,
>>>
>>>             I'm using Kamailio 3.1.5 in front of asterisk servers.
>>>             Kamailio handles all the SIP registrations. Calls from
>>>             SIP phones are forwarded to asterisks and then dialled
>>>             out to Kamailio.
>>>
>>>             root at SBCserver:~# kamailio -V
>>>             version: kamailio 3.1.5 (x86_64/linux) 76fff5
>>>             flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS,
>>>             TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST,
>>>             DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
>>>             DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
>>>             USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR,
>>>             USE_DST_BLACKLIST, HAVE_RESOLV_RES
>>>             ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
>>>             MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535,
>>>             PKG_SIZE 4MB
>>>             poll method support: poll, epoll_lt, epoll_et, sigio_rt,
>>>             select.
>>>             id: 76fff5
>>>             compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
>>>             root at SBCserver:~#
>>>
>>>
>>>             Problem:
>>>             When call is initiated from a softphone and is in
>>>             ringing phase, CANCEL just don't work. I've done some
>>>             initial debugging and the following piece of code in
>>>             main route is failing.
>>>
>>>             # CANCEL processing
>>>             if (is_method("CANCEL"))
>>>             {
>>>                  xlog("L_NOTICE","$rm from $fu (IP:$si:$sp)
>>>             ---CAPTURED IN MAIN---\n");
>>>                  if (t_check_trans()){
>>>                     t_relay();
>>>                     xlog("L_NOTICE","$rm from $fu (IP:$si:$sp)
>>>             ---CHECK TRANS TRUE---\n");
>>>                  }
>>>                  xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) ---CHECK
>>>             TRANS FALSE---\n");
>>>                  exit;
>>>             }
>>>
>>>             Also the CANCEL fails the has_totag() condition !
>>>
>>>             The same Call CANCEL scenario works fine for any client
>>>             on Public IP !
>>>
>>>             Hope to get some pointers for the solution.
>>>
>>>             Regards,
>>>             Sammy.
>>>
>>>
>>>
>>>
>>>         _______________________________________________
>>>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>         sr-users at lists.sip-router.org  <mailto: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
>>         Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat
>>         http://linkedin.com/in/miconda  -- http://twitter.com/miconda
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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
> Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat
> http://linkedin.com/in/miconda  -- http://twitter.com/miconda
>
>
> _______________________________________________
> 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
Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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