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

Eric Wieling EWieling at nyigc.com
Wed Sep 14 10:14:03 CDT 2011


If I read Kevin's post correctly, his statement applies to ALL echo cancellers, not just software EC.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gustavo Santos
Sent: Wednesday, September 14, 2011 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] High delay from Asterisk as PSTN simulator

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
	
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-- 
Atenciosamente,
Gustavo Santos.




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