Hello, I got to about 100 simultaneous calls, without touching rtps.
David On Thu, Jun 4, 2009 at 3:48 PM, Brett Nemeroff <[email protected]> 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 < > [email protected]> 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 < >> [email protected]> 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 >>> > >>> ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > Users mailing list >>> > [email protected] >>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > >>> >>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >
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