[OpenSIPS-Users] opensips 1.6.4 load balancing performance
Iulian Macare
iulian.macare at gmail.com
Wed Mar 7 10:40:41 CET 2012
I have installed Opensips 1.7.2 on debian and I have the same problem.
Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18883]: sending call
with callid 3817a6411cec76292a8d1c2811ad7def at 82.77.252.47 and ruri
sip:0103629834 at 82.77.252.40 to sip:82.77.252.48
Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18882]: sending call
with callid 3817a6411cec76292a8d1c2811ad7def at 82.77.252.47 and ruri
sip:0103629834 at 82.77.252.40 to sip:82.77.252.39
Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18880]: sending call
with callid 3817a6411cec76292a8d1c2811ad7def at 82.77.252.47 and ruri
sip:0103629834 at 82.77.252.40 to sip:82.77.252.48
Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18880]: sending call
with callid 3817a6411cec76292a8d1c2811ad7def at 82.77.252.47 and ruri
sip:0103629834 at 82.77.252.40 to sip:82.77.252.48
The called number is 0103629834 sent both to 82.77.252.39 and
82.77.252.48 many times
In the configuration file I have
if (uri=~"^sip:0[1-9][0-9]*@") {
load_balance("1","pstn");
xlog("sending call with callid $ci and ruri $ru to $du\n");
t_on_failure("1");
if (!t_relay())
{ sl_reply_error();
};
exit;
}
Destination:: sip:82.77.252.39 id=3 group=1 enabled=yes auto-re=on
Resource:: pstn max=200 load=36
Destination:: sip:82.77.252.35 id=1 group=1 enabled=yes auto-re=on
Resource:: pstn max=200 load=36
Destination:: sip:82.77.252.48 id=6 group=1 enabled=yes auto-re=on
Resource:: pstn max=200 load=36
The number 0103629834 I don't see it in opensips in accounting ( I use
CDRTool ) and I also don't see it in any logs on 82.77.252.39 or
82.77.252.48 ( wich are Asterisks )
opensips 1.6.4 load balancing performance
On Tue, Mar 15, 2011 at 6:46 PM, Iulian Macare <iulian.macare at gmail.com> wrote:
> I see also something like this:
>
> Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4547]: sending call with
> callid 3b77e3325a8c75f8571c4fcf37a8ba3c at 192.168.52.10 and ruri
> sip:0758553307 at 192.168.52.20 to sip:192.168.254.241
> Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4539]: sending call with
> callid 3b77e3325a8c75f8571c4fcf37a8ba3c at 192.168.52.10 and ruri
> sip:0758553307 at 192.168.52.20 to sip:192.168.254.241
> Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4556]: sending call with
> callid 3b77e3325a8c75f8571c4fcf37a8ba3c at 192.168.52.10 and ruri
> sip:0758553307 at 192.168.52.20 to sip:192.168.254.242
>
> Same call 3 times.. 2 to one server and another time to the other
>
> On Tue, Mar 15, 2011 at 12:10 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>>
>> Hi Iulian,
>>
>>
>>
>> Related to the second issue (with 2 logs for a single call), I suspect it
>> may be a retransmission issue. To be sure, print in your xlog the callid
>> also (to be sure it is the same request) . Var for callid is $ci.
>>
>> Regards,
>> Bogdan
>>
>>
>> Iulian Macare wrote:
>>>
>>> Hello
>>>
>>>
>>> I have installed OpenSips 1.6.4 on CentOS 5.5 32bit with load balancing &
>>> mysql support ; I want to balance calls to 2 asterisk servers . I am sending
>>> traffic to opensips from 1 x gnudialer & 1 x vicidial ( so from predictive
>>> dialers ). Situation is like this:
>>>
>>>
>>>
>>> +----+----------+--------------------------+-----------+------------+-------------+
>>> | id | group_id | dst_uri | resources | probe_mode |
>>> description |
>>>
>>> +----+----------+--------------------------+-----------+------------+-------------+
>>> | 1 | 1 | sip:192.168.254.241:5060 <http://192.168.254.241:5060>
>>> | pstn=300 | 0 | |
>>> | 2 | 1 | sip:192.168.254.242:5060 <http://192.168.254.242:5060>
>>> | pstn=300 | 0 | |
>>>
>>>
>>> +----+----------+--------------------------+-----------+------------+-------------+
>>>
>>> 600 channels in total , and I send around 500 channels ; OpenSips drops a
>>> lot of calls; By drop I mean a call that is not sent to one of those 2
>>> asterisk servers that I have.
>>>
>>> The code for balancing in this situation is:
>>>
>>> if (uri=~"^sip:0[1-9][0-9]*@") {
>>> load_balance("1","pstn");
>>> xlog("sending call $ru to $du\n");
>>> t_relay();
>>> exit;
>>> }
>>>
>>> ! An important thing to say is that in /var/log/messages I see the
>>> specific number that is sent to 192.168.254.241 for example ; So the
>>> parameter xlog("sending call $ru to $du\n"); works ; The problem is that in
>>> logs on 192.168.254.241 that number never arrives in asterisk logs ; In logs
>>> of vicidial & gnudialer I see it like congestion.
>>>
>>> If I do something like this:
>>>
>>>
>>> +----+----------+---------------------+-----------+------------+-------------+
>>> | id | group_id | dst_uri | resources | probe_mode |
>>> description |
>>>
>>> +----+----------+---------------------+-----------+------------+-------------+
>>> | 1 | 1 | sip:192.168.254.241 | pstn=150 | 0 |
>>> |
>>> | 2 | 2 | sip:192.168.254.241 | pstn=150 | 0 |
>>> |
>>> | 3 | 1 | sip:192.168.254.242 | pstn=150 | 0 |
>>> |
>>> | 4 | 2 | sip:192.168.254.242 | pstn=150 | 0 |
>>> |
>>>
>>> And I split opensips balancing in 2
>>>
>>>
>>> if(src_ip==192.168.3.10 )
>>> {
>>> load_balance("1","pstn");
>>> xlog("sending call to $du\n");
>>> t_relay();
>>> exit;
>>> };
>>>
>>> if(src_ip==192.168.3.11 )
>>> {
>>> load_balance("2","pstn");
>>> xlog("sending call to $du\n");
>>> t_relay();
>>> exit;
>>> };
>>>
>>>
>>> and by doing this I get the same numbers of channels on opensips ( around
>>> 500 channels ) but I am splitting in 2 groups of load balancing; It can
>>> process all the calls.
>>>
>>> Another question that I saw is that when I make a single call to opensips
>>> and I involve load balancing in /var/log/messages I get 2 times the same
>>> message .. just like it send 2 time to asterisk server the call .. but on
>>> asterisk I receive only one time.
>>>
>>> Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to
>>> sip:192.168.254.241
>>> Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to
>>> sip:192.168.254.241
>>>
>>>
>>> Isn't load balancing fast enough the process the calls made by predictive
>>> dialers, when over 300 channels is sent .. ? Or I have some mistakes made .
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS eBootcamp - 28th February 2011
>> OpenSIPS solutions and "know-how"
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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