If I read Kevin's post correctly, his statement applies to ALL echo cancellers, not just software EC.
-----Original Message----- From: [email protected] [mailto:[email protected]] 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 <[email protected]> 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: [email protected] | SIP: [email protected] | 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. -- _____________________________________________________________________ -- 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
