[SR-Users] dropped calls after 38 seconds openSER

Stoyan Mihaylov stoyan.v.mihaylov at gmail.com
Tue Apr 17 21:48:09 CEST 2012


I used wireshark on all interfaces and there I saw reentrance of ACK and
BYE.
Then I spent lot of time to find how to go around.

On Tue, Apr 17, 2012 at 10:41 PM, Saul Waizer <saulwaizer at gmail.com> wrote:

> Daniel, i got a fresh install kamailio 3.2.0 running on ubuntu per your
> suggestion. rtpproxy is running as well. My nathelper looks like this:
>
> # ----- nathelper params -----
> modparam("nathelper", "natping_interval", 3)
> modparam("nathelper", "ping_nated_only", 0)
> modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
> modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org")
> # SW changes
> modparam("nathelper", "natping_processes", 3)
> modparam("nathelper", "sipping_bflag", 7)
> modparam("nathelper", "sipping_method", "INFO")
>
> I also ran "kamctl fifo nh_enable_ping 1" on the console.
>
> Calls drop at *exactly 30* seconds, I dont see anything obvious in the
> logs. Any suggestions? Anyone?
>
> On Mon, Apr 16, 2012 at 2:51 PM, Saul Waizer <saulwaizer at gmail.com> wrote:
>
>> Daniel,
>>
>> thank you for the suggestion, I followed the tutorial and got it up and
>> running, however I am still experiencing the same issue, dropped calls
>> after 30 seconds. Furthermore the output of tshark -i eth0 -R sip does not
>> show any errors such as too many hops, everything looks very clean but I
>> still cant get more than 30 seconds out.
>>
>> I have a default config with mysql enabled and nat. rtproxy is running as
>> well.
>>
>> Any help is greatly appreciated.
>>
>>
>> On Mon, Apr 16, 2012 at 1:31 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> ngrep trace (I haven't seen any yet in the thread) of such call can help
>>> seeing if Record-/Route and Contact headers are properly set and maintained
>>> during the call. It might be a broken RR handling in a device or a wrong
>>> update of contact address.
>>>
>>> On the other hand, in the config, I saw presence being loaded -- if you
>>> need that, then start with kamailio 3.2.x, here is a tutorial:
>>>
>>> http://www.kamailio.org/wiki/install/3.2.x/git
>>>
>>> It comes with a default config file where is very easy to enable nat
>>> traversal as well as presence handling -- just read the top of config file
>>> and add the appropriate #!define directives.
>>>
>>> 1.3.x is anyhow way too old...
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>> On 4/16/12 7:16 PM, Stoyan Mihaylov wrote:
>>>
>>> I am using nathelper and rtpproxy.
>>> We had no serious problems with those modules.
>>> My idea was to overwrite wrong addresses in ACK and BYE packages,
>>> because they kept reentering kamailo, instead of being forwarded where they
>>> have to go.
>>>
>>> By the way - you can add:
>>>  modparam("htable", "htable", "forw=>size=8;autoexpire=7200;")
>>> modparam("htable", "htable", "mustforw=>size=8;autoexpire=7200;")
>>>
>>>  On Mon, Apr 16, 2012 at 8:03 PM, Saul Waizer <saulwaizer at gmail.com>wrote:
>>>
>>>> Thank you Stoyan
>>>> *
>>>> *
>>>> I added the routing config as you suggested but it still drops the call
>>>> after 30 seconds.
>>>>
>>>>  Any other suggestions?
>>>>
>>>>  Note: my nathelper module is commented out because I can't get it to
>>>> work, here is my mod list:
>>>>
>>>>   ###### Modules Section ########
>>>>
>>>>  #set module path
>>>> mpath="/usr/lib/openser/modules/"
>>>>
>>>>  /* uncomment next line for MySQL DB support */
>>>> loadmodule "mysql.so"
>>>> loadmodule "sl.so"
>>>> loadmodule "tm.so"
>>>> loadmodule "rr.so"
>>>> loadmodule "maxfwd.so"
>>>> loadmodule "usrloc.so"
>>>> loadmodule "registrar.so"
>>>> loadmodule "textops.so"
>>>> loadmodule "mi_fifo.so"
>>>> loadmodule "uri_db.so"
>>>> loadmodule "uri.so"
>>>> loadmodule "xlog.so"
>>>> loadmodule "acc.so"
>>>>
>>>>  loadmodule "auth.so"
>>>> loadmodule "auth_db.so"
>>>>
>>>>  #loadmodule "domain.so"
>>>> loadmodule "presence.so"
>>>> #loadmodule "presence_xml.so"
>>>>
>>>>  # !! Nathelper
>>>> #loadmodule "nathelper.so"
>>>> #loadmodule "nat_traversal.so"
>>>> #loadmodule "rtpproxy.so"
>>>> #loadmodule "dialog.so"
>>>>
>>>>  Thank You
>>>>
>>>>
>>>>  On Mon, Apr 16, 2012 at 12:03 PM, Stoyan Mihaylov <
>>>> stoyan.v.mihaylov at gmail.com> wrote:
>>>>
>>>>> Some time ago, I had similar problem.
>>>>> This was my solution:
>>>>> if(is_method("INVITE")){
>>>>>  ds_select_dst("1","4");
>>>>>  $sht(forw=>$ft)=$du;
>>>>>  sl_send_reply("100","Trying");
>>>>>  route(RELAY);
>>>>>  exit();
>>>>> }
>>>>>
>>>>>  if ( is_method("ACK|BYE") ) {
>>>>>  if ( t_check_trans() ) {
>>>>>  t_relay();
>>>>>  exit;
>>>>> } else {
>>>>>  if(($sht(forw=>$ft))=~$td){
>>>>>  $du=$sht(forw=>$ft);
>>>>>  }else if((($td=="sip.mydomain.com
>>>>> ")||($td=="ip.of.my.domain"))&&($si=="ip.of.my.domain")){
>>>>>  $du=$sht(forw=>$ft);
>>>>>  }
>>>>>  t_relay();
>>>>>  exit;
>>>>>
>>>>>>
>>>>>>  I am new to opensips so I am not too familiar with the routing
>>>>>> logics, a google search on that error suggests that there is a problem with
>>>>>> the route config where its creating a loop exhausting the Max Hops. The way
>>>>>> I configured my clients uses my server as a proxy.
>>>>>>
>>>>>>  Any help is greatly appreciated!
>>>>>>
>>>>>> On Mon, Apr 16, 2012 at 1:59 AM, davy van de moere <
>>>>>> davy.van.de.moere at gmail.com> wrote:
>>>>>>
>>>>>>> 38 seconds sounds pretty close to 30 seconds. Could those 38 seconds
>>>>>>> be a 30 seconds after the actual answer packet?
>>>>>>>
>>>>>>>  If so, you might want to look at ACK and OK packets not arriving
>>>>>>> correctly because of NAT, wrong IP selection in openser , etc...
>>>>>>>
>>>>>>>  A simple tshark might help you out to debug from a higher
>>>>>>> perspective : tshark -i eth0 -R sip
>>>>>>>
>>>>>>>  good luck!
>>>>>>>
>>>>>>> Op 16 april 2012 05:11 schreef Saul Waizer <saulwaizer at gmail.com>het volgende:
>>>>>>>
>>>>>>>>  Greetings list,
>>>>>>>>
>>>>>>>
>>>
>>
>
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