[OpenSIPS-Users] OpenSIPS 1.5.3, Load balancer module and open transactions

[Bogdan-Andrei Iancu]

Hi Gavin,

I just some small extension on 1.6 branch to print more info for the 
dialogs. So, right now, with dlg_list_ctx function, you can see the 
profiles the dialog belongs to.

The profiles used by LB can be easily spotted by the "lbX" prefix in name.

So list all dialogs with context and see which ones are counted by LB 
(based on profile) - the profile name and value will give you more info 
about destination and resource.

Best regards,
Bogdan

Gavin Henry wrote:
> On 6 December 2010 11:04, Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote:
>   
>> Hi Gavin,
>>
>> Do you think it will be helpful for you to "see" (from OpenSIPS) which are
>> the these ghost calls ? (I can describe a procedure to get such a listing).
>>     
>
> Yes, that would be great. We have tracked them down and working with
> the testing vendor why they didn't send us an ACK.
>
>   
>> Now, about what to do to prevent...More or less is about detecting so called
>> ghost calls....and you have several options:
>>
>>   1) callee/caller specific - if you are in control of one to the end points
>> (like you do LB over a bunch of Asterisk servers) :
>>     
>
> We do. Our OpenSIPS LBs also do the registration and have rtpproxy on them.
>
>   
>>           A) most of the media devices (PBX, GW, etc) implements media
>> timeout - if no RTP is recevied, they fire a BYE, so OpenSIPS will receive
>> this BYE and free the call
>>     
>
> Our media gateways (Asterisk) did do this, but OpenSIPS never got them
> for some reason.
>
>   
>>           B) as A), but using signalling, SST (when the call goes via
>> opensips, force SST on callee side). The detection will be done by the media
>> server, which will send a BYE
>>     
>
> OK, this could be good to compliment the no media.
>
>   
>>   2) opensips specific - as LB in the middle, you can check the health of
>> the dialog by:
>>            A) media level - both RTPproxy and mediaproxy offers RTP timeout
>> events (but this approach will require media relaying for all calls)
>>     
>
> OK, we relay media for NAT here. That will be seperate later.
>
>   
>>            B) signalling level - opensips cannot do much here at the moment
>> (like generating in dialog request for probing purposes)
>>     
>
> OK.
>
>   
>>
>> Regards,
>> Bogdan
>>
>> Gavin Henry wrote:
>>     
>>> On 5 December 2010 01:07, Advantia VoIP Systems <info at advantia.ca> wrote:
>>>
>>>       
>>>> Can you verify that a BYE is sent on one UAC and and received on the
>>>> other?
>>>>
>>>>         
>>> We're checking. It looks like it's the technical testing calls as part
>>> of the ITSPA Awards 2011 - http://www.itspaawards.org.uk/
>>>
>>> We've got traces on as we speak. Even so, how can we handle not
>>> receiving BYEs on the lb? We're open to DOS, if we get lots of these
>>> our LBs will fill up. We have pike and ratelimit on anyway.
>>>
>>> Thanks.
>>>
>>>
>>>