[SR-Users] Kamailio direct interconnectivity with PRI

[David J]

Well any gateway software that does protocol conversion is going to be more
overhead than pure sip.

Something has to do the protocol conversion.

Typically if you want to avoid using asterisk etc. The you can just buy a
true media gateway like audio codes which does exclusively that. Usually
the hardware solution has better performance
On Oct 11, 2012 9:18 AM, "SamyGo" <govoiper at gmail.com> wrote:

> Hello Sir,
>
> With due respect, the only answer for your question  I have is that, I
> think that the capacity handling of currently popular media servers is less
> than that of Kamailio.
>
> I agree that all of current media servers are getting better and stronger
> but why should Kamailio get bottlenecked by Media-Servers in front of it.
>
> By the way I completely understand the current option of installing
> Asterisk/FreeSWITCH/Yate and tell them to send calls to Kamailio or
> anything else. Proudly enough I've done such setups happily before. I just
> wanted to reduce a hop between kamailio and the PRIs.
>
> Thanks,
> Sammy
>
>
>
>
> On Thu, Oct 11, 2012 at 6:04 PM, Olle E. Johansson <oej at edvina.net> wrote:
>
>>
>> 11 okt 2012 kl. 13:57 skrev SamyGo <govoiper at gmail.com>:
>>
>> :) "Soon..." But Not Today.
>>
>> Not everyone can afford the Gateways. Thanks for the replies. I was
>> hoping maybe someone else be thinking of freeing the kamailio from
>> Asterisks or Freeswitchs when it comes to interconnecting with PSTN.
>>
>> Why? We have plenty of good media servers out there that can handle those
>> use cases. Few of them handle SIP correctly and include new SIP features
>> like presence in a proper way. Combining them is a good match.
>>
>> :-)
>>
>> /O
>>
>> Cheers
>>
>> Sammy
>>
>>
>> On Oct 11, 2012 4:09 PM, "Neill Wilkinson" <
>> neill.wilkinson at btinternet.com> wrote:
>>
>>> You might consider:
>>>
>>>
>>> http://sangoma.com/products/voip_gateways/netborder_software/netborder_express.html
>>>
>>> Then put Kamailio in front of that... Simple Gateway PRI -> SIP.
>>>
>>> Neill....;o)
>>>
>>> Aeonvista Ltd
>>> Opening Up New Ideas
>>>
>>>
>>> On 11 October 2012 11:43, SamyGo <govoiper at gmail.com> wrote:
>>>
>>>> Hi David,
>>>>
>>>> Thanks for this useful information.  What I don't like is that if I've
>>>> couple of Sangoma cards, each supporting 120 channels each,  is using any
>>>> of this B2BUAs in between the cards and Kamialio and get a limited capacity
>>>> application in front of Kamailio. !
>>>>
>>>> If there is any driver for kamailio where possibly use
>>>> media-proxy/rtpproxy for handling the PRI-Channels media and then
>>>> distribute my calls to media-servers i.e SMES/Asterisk/yate/FS/XYZ
>>>>
>>>> PRIs <===>> Driver+Kamailio <=====> Asterisks/FreeSWITCHs
>>>>
>>>> Just want to know if technically any such driver program is doable or
>>>> not !
>>>>
>>>> Thanks,
>>>> Sammy
>>>>
>>>>
>>>> On Thu, Oct 11, 2012 at 3:31 PM, David J <david at styleflare.com> wrote:
>>>>
>>>>> Asterisk yate or free switch.
>>>>>
>>>>> You need something as a gateway between PRI and sip. Kamailio does not
>>>>> handle this conversion
>>>>> On Oct 11, 2012 6:24 AM, "SamyGo" <govoiper at gmail.com> wrote:
>>>>>
>>>>>>  Hello,
>>>>>>
>>>>>> I've a scenario in which I've to deploy a couple Sangoma PRI cards
>>>>>> with kamailio. What I wish is that I've some drivers for this purpose and
>>>>>> so I don't ned to install FreeSWITCH or Asterisk in between the PRIs and
>>>>>> Kamailio.
>>>>>>
>>>>>> Kindly give any feedback on what are the possibilities and options.
>>>>>>
>>>>>> Thanks,
>>>>>> Sammy
>>>>>>
>>>>>>
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>>>>>>
>>>>>>
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>>>>>
>>>>
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>
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