[Freeswitch-users] Voice lag in conference

[Anthony Minessale]

The problem comes from the timing of certain phones during the capture of
audio actually clocked slightly faster than what it advertises.
Try the latest trunk with all the defaults in your sip profile as we have
tried to make the defaults deal with this automatically.


On Tue, Jun 16, 2009 at 12:51 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> 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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>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

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