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.
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