[asterisk-users] High delay from Asterisk as PSTN simulator
Gustavo Santos
gustavo at voip.ufrj.br
Wed Sep 14 14:37:23 CDT 2011
I'm trying to simulate the situation:
SIP <----> Asterisk <-------> PSTN
In this case 16 ms works?
I've read in voip-info: "Simplistically, you'd need a "tail circuit" (the
distance between your echo canceller and the source of the echo) of over
2500 miles to acheive an echo path of 30ms [...] Asterisk's default of
128taps will therefore handle echo paths of up to 16ms, and is therefore
probably good for most things.".
2011/9/14 Kevin P. Fleming <kpfleming at digium.com>
> On 09/14/2011 10:14 AM, Eric Wieling wrote:
>
>> If I read Kevin's post correctly, his statement applies to ALL echo
>> cancellers, not just software EC.
>>
>
> That's correct; regardless of whether the EC runs on a DSP or on the host
> CPU (or is a device in the path of an T1/E1 circuit), if it can only handle
> a 16ms window for echo cancellation, then having a packetized 'hop' in the
> audio path is likely to make it unable to do its job at all.
>
> Echo cancellers used on networks where packetized audio is transported
> through at least one hop need to be configured for (at least) 64ms windows,
> if not 128ms.
>
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
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Atenciosamente,
Gustavo Santos.
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