[OpenSIPS-Users] H323

[David Villasmil]

Hello,

I got to about 100 simultaneous calls, without touching rtps.

David



On Thu, Jun 4, 2009 at 3:48 PM, Brett Nemeroff <brett at nemeroff.com> wrote:

> David,What capacities are you able to achieve with yate and h323? roughly?
>
>
> On Thu, Jun 4, 2009 at 7:15 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>>
>>      yate is great for h.323<->SIP, it even has a config on its web to do
>> it. I've used it a lot.
>>
>> have fun!
>>
>>
>>
>>
>> On Mon, Jun 1, 2009 at 12:55 PM, Bogdan-Andrei Iancu <
>> bogdan at voice-system.ro> wrote:
>>
>>> Wayne Scholar wrote:
>>> > A bit more info.
>>> >
>>> > We use opensips as a gateway to know about all our asterisk servers
>>> > behind it.  We do not use it for media just for signaling and so we
>>> > don't have to register IPs with our carriers other than our gateway.
>>> > The opensips will dist the calls across our asterisk servers.
>>> >
>>> > I am looking to build the same setup with h323 protocol.  So i don't
>>> > know if i need a gateway or gatekeeper or what software can perform
>>> > this function.  freeswitch is a lot like asterisk as a read it?
>>>
>>> I think yate has some nice h323 support - you may look around to see if
>>> it can do what you need.
>>>
>>> Regards,
>>> Bogdan
>>> >
>>> > Wayne
>>> >
>>> > On Fri, 2009-05-29 at 15:28 -0500, Brett Nemeroff wrote:
>>> >> freeswitch
>>> >
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>>
>>
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