[SR-Users] CANCEL not matching INVITES !

Daniel-Constantin Mierla miconda at gmail.com
Fri Dec 2 18:55:23 CET 2011


Hello,

On 12/2/11 5:24 AM, Sammy Govind wrote:
> Hello again,
>
> You were right, as soon as I made changes in asterisk SIP profile for 
> the Kamailio proxy server and stopped the 401 Auth from Asterisk to 
> Kamailio the CANCELS started to work fine.
well, the 401 from asterisk is ok from specs point of view (although 
many phones don't work with many challenges), but this case revealed 
some bugs in asterisk as well as in xlite, both of them had misbehavior.

Cheers,
Daniel

>
> So the SIP flow now is:
>
> - invite from phone to kamailio
> - kamailio asks for authentication - 407
> - ack
> - invite with credentials, kamailio forwards to asterisk
> - asterisk starts processing the invite and call can be cancelled now.
>
>
> Thanks alot
>
> --
>
> Best Regards,
> Sammy.
>
> On Thu, Dec 1, 2011 at 12:01 PM, Sammy Govind <govoiper at gmail.com 
> <mailto:govoiper at gmail.com>> wrote:
>
>     Hey Daniel,
>
>     I've exactly followed your point, I'll try some stuff on asterisk
>     server to stop asking for 401 Auth to Kamailio., maybe this will
>     eliminate the need for another INVITE with authentication params.
>
>     But one thing which just makes me curious is that a soft phone
>     directly coming from a Public IP is always able to successfully
>     CANCEL the call.
>
>     Anyway I'll use some brain of mine on this and let you know what
>     resolved it, or what I'm missing.
>
>     Thanks,
>     Sammy
>
>
>     On Wed, Nov 30, 2011 at 5:47 PM, Daniel-Constantin Mierla
>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>         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  <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  <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

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