[Asterisk-Users] Why echo occurs

[Steven Critchfield]

On Fri, 2005-02-11 at 15:35 +1100, Eric Bishop wrote:
> 2. Is only a problem in 2-wire technologies (ie analog and BRI ISDN lines)?

It is just an analog problem. That is why a BRI can actually transmit
direct digital data instead of audio data. 

> 3. Where exactly is the slowdown occuring? For example take my Supira
> 3000 as a case in point. It takes no longer for the PSTN signal to
> reach the Sipura's FXO port than it does my $5 handset. Going from the
> other end it takes no longer for the SIP signal to reach to the
> Sipura's ethernet port than it does any other IP phone. So logically
> the slowdown is happening as Sipura converts the PSTN signal to SIP
> and so forth. Is it just that the Sipura/TDM400 etc. have a too slow
> conversion CPU. Would a faster digital to analogue audio converter
> "fix" the the problem?

Steve Underwood pointed out that most Telco equipment has a max delay of
3 samples. On your Sipura, you will have a initial packetization delay
of 160 samples. And that is if there isn't any compression work time.
You can bet that no matter what it never takes longer than half the time
to receive the samples to compress them as it is likely a symmetrical
codec and the other half of the time is decoding the incoming stream
too.

Next, if the hop from the Sipura to your PBX takes half a millisecond,
you still add the equivalent of 4 samples of delay. 

If you have asterisk doing any work on the link, you add more delay
dependent on speed of system and amount of work done. 

If you route very far, you add more delay. For example, on a point to
point T1 link where it only traveled 20 miles or so and was not
congested, I still saw another 3-4ms of delay for a ping. So figure
1.5-2 ms for each side of the hop. So figure another 12-16 samples of
delay.

Of course out my cable modem and up to my office asterisk machine is
showing 46.9ms average round trips. So easily I am looking at a full
voice packet in transit while the next one is being created. And I'm
only 14 hops away all on the AT&T network.

So if normal toll quality calls have no more than 3 samples delay, you
are looking at a minimum of 164 samples delay on VoIP and possibly more
than 330-340 samples. 110 times slower than what the telcos would use.

-- 
Steven Critchfield <critch at basesys.com>