[asterisk-users] End to End delay calculation

[Steve Edwards]

On Sun, 22 Nov 2009, capricorn 80 wrote:

> I am looking to calculate the end-to-end delay between two soft 
> phone/hard phone. I have asterisk server and configured ntp server on 
> the same machine and synchronized it with ntp pool.  I have seen that 
> Wireshark can be used to check the jitter. But I am not sure how can i 
> calculate the end to end. May be this is not related to the mailing list 
> topic but please help me if anyone has some information.

A very long time ago, I made the mistake of letting a client listen (with 
a handset on each side of his head) to end-to-end delay.

This all of a sudden became a quest for the Holy Grail to quantify and 
reduce the delay.

I got a couple of RadioShack telephone recording interfaces, connected one 
to each endpoint. Then I connected the outputs to the left and right 
channels on a PC and recorded "tapping" on one of the handsets using 
Audacity. When I selected the interval between the "tap" and the "ping," 
Audacity would show the time in ms.

All very "old-school" but it worked and the client never questioned the 
"pretty pictures" on the computer screen.

Wireshark may be able to tell you how long it takes a packet to travel 
across your network, but what about the time from the network interface on 
the host until sound comes out the earpiece? How long does it take a SIP 
phone to take a packet off it's network interface, wiggle it through it's 
jitter buffer, transcode it, convert it to analog and deliver it to the 
earpiece?

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000