[Kamailio-Users] how to know if the next gw is up?

Daniel-Constantin Mierla miconda at gmail.com
Fri Feb 27 10:24:39 CET 2009


Please direct your questions to users mailing lists. Private messages of 
such type are not honored very fast, they usually end in my very long 
unread mails pool.

See the dispatcher module, maybe it helps you.

Thanks,
Daniel


On 02/27/2009 11:18 AM, BERGANZ François wrote:
> Hello,
>
> I have 1SER and 5 Asterisk.
> I need to choose the dst with my algo...
> But, I need to have my server list and choose a server which is online.
> The best thing could be to have OPTIONS send to my asterisk and write responses in the database!
>
> Have you an idea?
>
>
> Cordialement,
> BERGANZ François
>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
>
> -----Message d'origine-----
> De : users-bounces at lists.kamailio.org [mailto:users-bounces at lists.kamailio.org] De la part de Daniel-Constantin Mierla
> Envoyé : vendredi 27 février 2009 09:10
> À : c_lougher at yahoo.co.uk
> Cc : users at lists.kamailio.org
> Objet : Re: [Kamailio-Users] Kamailio Newb questions
>
>
>
> On 02/27/2009 12:07 AM, carl Lougher wrote:
>   
>> So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy?
>>   
>>     
> nathelper/rtpproxy does rtp relay on server, so the rtp goes: [caller] 
> === [rtpproxy] === [callee]
>
> In case of dealing with symmetric NAT, it is the only feasible way now 
> to make it work.
>
> If STUN is used, the rtp goes: [caller] === [callee] . But does not work 
> for symmetric nat.
>
> The best is to combine both of them so you get as less as possible RTP 
> relaying on server.
>
> Cheers,
> Daniel
>
>   
>> What are you using rtpproxy for that is different than stun?
>>
>>
>> --- On Thu, 26/2/09, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>>
>>   
>>     
>>> From: Daniel-Constantin Mierla <miconda at gmail.com>
>>> Subject: Re: [Kamailio-Users] Kamailio Newb questions
>>> To: c_lougher at yahoo.co.uk
>>> Cc: users at lists.kamailio.org
>>> Date: Thursday, 26 February, 2009, 12:27 PM
>>> On 02/26/2009 01:19 PM, carl Lougher wrote:
>>>     
>>>       
>>>> Thanks for that. So does it mean by using rtpproxy you
>>>>       
>>>>         
>>> will therefore carry all the rtp streams through that server
>>> yes, that is the role of RTPProxy - to proxy the RTP
>>> streams, therefore those go via the server.
>>>
>>> If you want end-to-end RTP stream, then look at STUN, if
>>> the phones are not behind symmetric nat, it can help.
>>>
>>>     
>>>       
>>>>  or can it be redirected to the sip provider from the
>>>>       
>>>>         
>>> endpoint?
>>>     
>>>       
>>>> Also how do you  put the kamailio server in the
>>>>       
>>>>         
>>> equation? Do you set it up as an external proxy for the
>>> clients or do you register the clients to it then just use
>>> asterisk for the media/vmail etc?
>>>     
>>>       
>>>>   
>>>>       
>>>>         
>>> I do everything in kamailio but the media services which i
>>> do with asterisk (vmail, ivr, ...) - authentication,
>>> registration, call routing is done in kamailio.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>     
>>>       
>>>> --- On Thu, 26/2/09, Daniel-Constantin Mierla
>>>>       
>>>>         
>>> <miconda at gmail.com> wrote:
>>>     
>>>       
>>>>   
>>>>       
>>>>         
>>>>> From: Daniel-Constantin Mierla
>>>>>         
>>>>>           
>>> <miconda at gmail.com>
>>>     
>>>       
>>>>> Subject: Re: [Kamailio-Users] Kamailio Newb
>>>>>         
>>>>>           
>>> questions
>>>     
>>>       
>>>>> To: c_lougher at yahoo.co.uk
>>>>> Cc: users at lists.kamailio.org
>>>>> Date: Thursday, 26 February, 2009, 9:16 AM
>>>>> Hello,
>>>>>
>>>>> On 02/26/2009 12:59 AM, carl Lougher wrote:
>>>>>     
>>>>>         
>>>>>           
>>>>>> Howdy,
>>>>>> I'm trying to remove the media/rtp streams
>>>>>>           
>>>>>>             
>>> from an
>>>     
>>>       
>>>>>>       
>>>>>>           
>>>>>>             
>>>>> asterisk server for natted users so would like to
>>>>>         
>>>>>           
>>> know if
>>>     
>>>       
>>>>> this is possible with kamailio.
>>>>>     
>>>>>         
>>>>>           
>>>>>>         
>>>>>>           
>>>>>>             
>>>>> yes it is possible. nathelper+rtpproxy is the
>>>>>         
>>>>>           
>>> option I use
>>>     
>>>       
>>>>> and prefer because of flexibility and
>>>>>         
>>>>>           
>>> performances. You can see an
>>>     
>>>       
>>>>> example at:
>>>>>
>>>>>         
>>>>>           
>>> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
>>>     
>>>       
>>>>>     
>>>>>         
>>>>>           
>>>>>> Qu's:
>>>>>> What is the best option?
>>>>>> rtpproxy/mediaproxy?
>>>>>> nathelper?
>>>>>>
>>>>>> If i use kamailio to achieve this does it mean
>>>>>>           
>>>>>>             
>>> that i
>>>     
>>>       
>>>>>>       
>>>>>>           
>>>>>>             
>>>>> still have to carry the rtp streams through the
>>>>>         
>>>>>           
>>> kamailio
>>>     
>>>       
>>>>> server instead?
>>>>>     
>>>>>         
>>>>>           
>>>>>>         
>>>>>>           
>>>>>>             
>>>>> through the rtpproxy server, which can be located
>>>>>         
>>>>>           
>>> on same
>>>     
>>>       
>>>>> or different machine than kamailio.
>>>>>
>>>>>     
>>>>>         
>>>>>           
>>>>>> Also will i need to change the logon info for
>>>>>>           
>>>>>>             
>>> the
>>>     
>>>       
>>>>>>       
>>>>>>           
>>>>>>             
>>>>> clients so they now logon to kamailio then i just
>>>>>         
>>>>>           
>>> point
>>>     
>>>       
>>>>> registrar to asterisk?
>>>>>     
>>>>>         
>>>>>           
>>>>>> Can i use kamailio for sip trunks to asterisk
>>>>>>           
>>>>>>             
>>> and
>>>     
>>>       
>>>>>>       
>>>>>>           
>>>>>>             
>>>>> carry rtp and natted clients media streams rather
>>>>>         
>>>>>           
>>> than
>>>     
>>>       
>>>>> register to asterisk?
>>>>>     
>>>>>         
>>>>>           
>>>>>>         
>>>>>>           
>>>>>>             
>>>>> Yes, you can register to kamailio, see registrar
>>>>>         
>>>>>           
>>> and usrloc
>>>     
>>>       
>>>>> modules.
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>>     
>>>>>         
>>>>>           
>>>>>> Many thanks,
>>>>>> Taff..
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>       
>>>>>>
>>>>>>           
>>>>>>             
>>> _______________________________________________
>>>     
>>>       
>>>>>> Kamailio (OpenSER) - Users mailing list
>>>>>> Users at lists.kamailio.org
>>>>>>
>>>>>>       
>>>>>>           
>>>>>>             
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>>     
>>>       
>>>>>    
>>>>>         
>>>>>           
>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>     
>>>       
>>>>>     
>>>>>         
>>>>>           
>>>>>>         
>>>>>>           
>>>>>>             
>>>>> -- Daniel-Constantin Mierla
>>>>> http://www.asipto.com
>>>>>     
>>>>>         
>>>>>           
>>>>       
>>>> _______________________________________________
>>>> Kamailio (OpenSER) - Users mailing list
>>>> Users at lists.kamailio.org
>>>>
>>>>       
>>>>         
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>>     
>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>     
>>>       
>>>>   
>>>>       
>>>>         
>>> -- Daniel-Constantin Mierla
>>> http://www.asipto.com
>>>     
>>>       
>>       
>>
>> _______________________________________________
>> Kamailio (OpenSER) - Users mailing list
>> Users at lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>
>>   
>>     
>
>   

-- 
Daniel-Constantin Mierla
http://www.asipto.com





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