[asterisk-users] High delay from Asterisk as PSTN simulator

[Gustavo Santos]

So any software echo canceller available in dahdi isn't good enough?

2011/9/13 Kevin P. Fleming <kpfleming at digium.com>

> On 09/13/2011 08:56 AM, Gustavo Santos wrote:
>
>> I'm trying to use Asterisk as a PSTN simulator to run performance tests
>> for echo cancellation algorithms. I'm using the following configuration:
>>
>> SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo()
>>
>> Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan
>> application.
>>
>> The problem is the high delay using this configuration: 20 ms only in
>> Asterisk 2. I've read the source code of chan_dahdi, and I think the
>> channel has a 20 ms "buffer" (160 samples). Algorithms like mg2 and kb1
>> are configured to accept 128 taps (16 ms), so 20 ms is too high.
>>
>> Someone knows how I can reduce the delay to at least 10 ms? Should I
>> change something in the source code?
>>
>
> 20 milliseconds is far from a 'high' (long) delay. Asterisk handles audio
> in packets, it does not directly switch TDM streams. As a result, there is
> always going to be (at least) the delay of one packet time for audio passing
> into Asterisk and back out via the Echo() application. This is unavoidable.
>
> An alternative solution would be to send a call into Asterisk2 and have it
> dial back to Asterisk1 (and then back to the originating endpoint) and
> bridge those two calls in Asterisk2; if both calls are on the same E1, then
> Asterisk will let the DAHDI hardware directly connect the two channels,
> resulting in a 1 or 2 millisecond delay.
>
> But realistically... configuring an echo canceller with only a 16ms window
> of operation is not very practical. Sending a call through *any* network
> element that packetizes the audio will result in a delay longer than 16ms.
>
> --
> 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
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>



-- 
Atenciosamente,
Gustavo Santos.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110914/e9b69763/attachment.htm>