[Asterisk-Users] SPA-841 "Decode Latency"?

[alan]

> Subject: Re: [Asterisk-Users] SPA-841 "Decode Latency"?

"Matias G." <listas_ast at reliable.com.ar> wrote:

> > Luki <lugosoft at gmail.com> wrote:
> >
> >> > Does anyone have any familiarity with "decode latency," specifically
> >> > with Sipura devices? Why would it take 200+ms to decode a 20ms RTP
> >> > packet? G.711u has existed for over 30 years, how hard could it be?

<snip>

> > Our current working theory, which we will test "soon", is that this may
> > be caused by periodic high levels of ARP broadcast traffic. I'm not
> > familiar with the hardware of these phones, and for most ethernet
> > devices they should ignore ARP with no performance effects. But if the
> > SPA-841 is set up in such a way that it eats CPU for the phone to
> > discard ARP packets, then this could be a problem for us.
> >
> > I'll keep you posted on what we find. If anyone has any insight into the
> > networking hardware the SPA-841 uses, I'd be interested in that.
>
> Alan,
> pls keep us informed on wether you find what is going wrong with this
> issue... thanks a lot.
>
> M.

Hello,

Although we don't have a definitive answer for why we were having the
problems we were having, we seem to have solved those problems.

Specifically:

We have taken portions of our SIP phone network, and completely
separated them from the rest of our network. The Asterisk server's
second ethernet port is dedicated for use with the SIP phones only, and
wiring from there to the phones is used only for phone SIP traffic.

Since the network was fully switched before, the only real effect this
has is to segment the broadcast domain. In terms of traffic reaching the
phones, the phones are no longer seeing any ARP packets (or other
broadcast packets, minimal) destined for the rest of the network.

This has pretty much 100% solved our sound issues, which manifested
themselves as "robot voice", buzzing, and dropout (silence).

This is a tremendous relief! Now, we just need to figure out how to
deploy two completely parallel networks and wiring, which kind of
defeats one of the original purposes of going with VOIP...

I haven't specifically checked on the "decode latency" value lately, for
a few reasons:

- I haven't had any audio issues to mark a good time to check on it.
  Previously, the decode latency was usually normal, and
  only rarely "very high", so a random sampling would probably not show
  me much useful anyway.
- Now that the phones are on their own network, it's a lot more
  difficult to get to the web configuration screen :) This is
  something I need to work on, but all of our phone configuration is now
  centrally provisioned so it's not a big deal in practice.

Thank you all for your insight,

Alan Ferrency
pair Networks, Inc.
alan at pair.com