[Freeswitch-users] Voice lag in conference

[Bradley Brashier]

I'm creating a conferencing product for use in a system with theoretically
several hundred concurrent calls. I'm using FreeSwitch to create this
product, but am not only new to FreeSwitch, but also the entire telecom
industry as well as Open Source projects in general (I'm a recovering BIOS
guy).

I've got a bare-bones conference up and running on the server, including a
handshake and a couple of features, and am using the default packages from
the current trunk, but I've noticed that voice lag is a pretty big issue.
Common lag times are several hundred milliseconds, and I've heard as long as
a second. It seems to be at least marginally specific to individual phones
-- certain phones have longer lag than others even on the same call.

My question is really about what my options are. Is this just a part of SIP?
Of conferencing? Of FreeSwitch? Are there things I can prune or slim down
that will help? Is this a common issue? If it's common, is it expected by
the marketplace?
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