[Asterisk-Dev] Re: How to measure delay in meetme?

[Steve Edwards]

On Mon, 29 Aug 2005, Tony Mountifield wrote:

> I've never found problems with delay when only Zap channels are involved;

Personally, I don't think anybody would notice in a real conversation, but 
you can notice it if you have a separate phone to each ear or if you 
listen on one handset and tap the table with the other.

> So Steve, your MeetMe conferences *only* involve Zap channels on your
> T1 connection? And it is in these that the boss complains of delay?
> With no VoIP channels involved?

Zap only.

Looking at the "show channel" for all involved, the Zap/N channels have 
Native, Write, Read of 68, 64, 64 and the Zap/pseudo channel shows 68, 4, 
4. Would there be anything gained by trying to get everybody to use the 
same codec?

Are there any gains to be made fiddling with echo, buffer, or jitter 
parameters?

I'd really like to be able to measure the delay. This would allow me to 
make a rational argument like "cell phones have a delay of x ms and 
nobody notices" or "research shows the limit of human perception is y ms."

This would also allow me to tell if any of the changes I make actually do 
anything :)

On Mon, 29 Aug 2005, Tony Mountifield wrote:

> In article <Pine.LNX.4.61.0508281750221.11671 at fs.sedwards.com>,
> Steve Edwards <asterisk.org at sedwards.com> wrote:
>> Thanks for the checklist.
>>
>>> 1. Use a good hardware timing source.  Zaptel card, or if ztdummy,
>>> enable USE_RTC (it's in the source in head, look for it)
>>
>> t100p
>
> So Steve, your MeetMe conferences *only* involve Zap channels on your
> T1 connection? And it is in these that the boss complains of delay?
> With no VoIP channels involved?
>
> I've never found problems with delay when only Zap channels are involved;
> it's always with VoIP channels.
>
> The audio mixing is done in the zaptel device driver in real time between
> the Zap channels that are involved, so the delay should be minimal. When
> VoIP is involved, each VoIP channel gets a proxy Zaptel Pseudo-channel and
> MeetMe has to copy the audio data both ways between the VoIP and Pseudo
> channels. It is here that the delays creep in.
>
> Cheers
> Tony
> -- 
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
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Thanks in advance,
------------------------------------------------------------------------
Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline           pagesteve at sedwards.com            Fax: +1-760-731-3000