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  #1 (permalink)  
Old 07-10-2007, 01:46 PM
Jeff Liebermann
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Default Re: Network requeriments of VoIP ?

landuchi <landuchi.2thiqu@no-mx.wirelessforums.org> hath wroth:

>Well, basically the title says it all.


Title? Oh, I guess I have to click on display header and check the
title in my newsreader. You could have repeated the subject in your
message, but it's more fun to make the reader go through a few hoops
to figure out what you're asking.

>If I set a wireless network for
>data and VoIP what would the requirements be?


That varies by the bandwidth available on your wireless network, by
the type of codec you're using (G.711, G.729, etc), and the number of
VoIP sessions. The bandwidth should not be a problem as even the
slowest connection speed (1Mbit/sec) will give you about 300Kbits/sec
thruput. VoIP usually requires 30-100Kbit/sec per call.

>What are the bandwith requirements of each VoIP call ?


Depends on the codec. See chart at:
<http://www.erlang.com/bandwidth.html>

>Whats is the maximum latency allowed for a VoIP call ?


That also varies with the codec. See:
<http://www.erlang.com/calculator/>
for various calculators. Be sure to use half-duplex for wi-fi.
This looks better:
<http://www.connect802.com/voip_bandwidth.php>

In general, less than 50 msec will work for most codecs. A properly
functioning Wi-Fi system will contribute less than 5msec latency for a
small indoor system. Less than 1 msec is not unusual. However, if
you start getting errors or inteference, the latency and packet loss
will climb. The rest of the latency budget is mostly tied up in your
broadband connections. Ping your ISP gateway. That's most of your
latency.

>Are there other requirements ?


Yes. Same requirement as usual necessary to answer any question:
1. What problem are you trying to solve?
2. What do you have to work with? (Equipment, software, versions).
3. What have you done so far and what happened? (for
troubleshooting).

--
Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558

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  #2 (permalink)  
Old 07-10-2007, 02:16 PM
Jeff Liebermann
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Default Re: Network requeriments of VoIP ?

Jeff Liebermann <jeffl@cruzio.com> hath wroth:

>>Whats is the maximum latency allowed for a VoIP call ?


>In general, less than 50 msec will work for most codecs.


Oops, I goofed. 50 msec of *jitter* is about the maximum. That's the
average variation in latency over a short period of time. Jitter has
a big effect on voice quality. The actual round trip latency can be
almost anything. VoIP will even sorta work over a satellite link that
has a minimum latency of 800 msec. It sounds weird, requires
considerable coordination, drives me nuts when someone tries it, but
does work. For example, a cellular voice connection has about 500msec
latency, which you can hear if you call yourself on a POTS phone, and
listen to the voice delay on the cell phone.

Wireless is a particularly bad problem for VoIP because of commonly
high packet loss. That doesn't cause much trouble for data
communications, but rips on VoIP quality. I use Skype on my VX6700
cell phone via 802.11b. It often sounds rather bad due to 2.4GHz
interference causing packet loss.

Look for low jitter, not low latency.

This covers the basics:
<http://expertanswercenter.techtarget.com/eac/blog/0,295203,sid63_tax303505,00.html>
There's some notes in there on jitter.

VoIP and Jitter tester:
<http://myvoipspeed.visualware.com>
I'm having a good day. 0.0% packet loss and 0.8msec jitter.


--
Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558

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  #3 (permalink)  
Old 07-11-2007, 10:10 PM
stephen
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Posts: n/a
Default Re: Network requeriments of VoIP ?

"Jeff Liebermann" <jeffl@cruzio.com> wrote in message
news:v14793156js2f3feu6h3atbi4c69nrcuq6@4ax.com...
> Jeff Liebermann <jeffl@cruzio.com> hath wroth:
>
> >>Whats is the maximum latency allowed for a VoIP call ?

>
> >In general, less than 50 msec will work for most codecs.

>
> Oops, I goofed. 50 msec of *jitter* is about the maximum. That's the
> average variation in latency over a short period of time. Jitter has
> a big effect on voice quality. The actual round trip latency can be
> almost anything. VoIP will even sorta work over a satellite link that
> has a minimum latency of 800 msec. It sounds weird, requires
> considerable coordination, drives me nuts when someone tries it, but
> does work. For example, a cellular voice connection has about 500msec
> latency, which you can hear if you call yourself on a POTS phone, and
> listen to the voice delay on the cell phone.


there is an ITU spec for this (there is for anything remotely to do with
voice or telecoms).

max "ear to mouth" latency 1 way of 150 mSec, jitter of 50 mSec, packet loss
of 1%.

Choice of codec doesnt change the overall delay budget, but more complex
ones and / or high compression use more for the code / decode scheme.

a typical voip system (say G.711, or 729) encodes samples of 10 mSec of
speech and puts 2 into a packet.

so you have around 30 mSec latency on the Tx end, a similar amount at the
reciever, and a playout buffer to cope with arrival jitter.

So 50 to 100 mSec to play with on the actual network delay...

bandwidth avries by codec and network overhead - Ethernet and 802.11
overhead is going to be significant as you have 50 small packets / sec to
wrap....... but still "only" 40 to 100 Kbps full duplex.
>
> Wireless is a particularly bad problem for VoIP because of commonly
> high packet loss. That doesn't cause much trouble for data
> communications, but rips on VoIP quality. I use Skype on my VX6700
> cell phone via 802.11b. It often sounds rather bad due to 2.4GHz
> interference causing packet loss.


Poor uplinks on a broadband link can hurt Voip pretty easily, but the usual
culprits seem to be simultaneous use by other protocols on a shared link, or
temporary congestion killing the delay out in the internet.

The worst characteristic is the way problems often only affect 1 direction,
so someone keeps talking obliviously even though the person at the other end
just gets distortion, clipping and the odd darlek.....
>
> Look for low jitter, not low latency.
>
> This covers the basics:
>

<http://expertanswercenter.techtarget...id63_tax303505,
00.html>
> There's some notes in there on jitter.
>
> VoIP and Jitter tester:
> <http://myvoipspeed.visualware.com>
> I'm having a good day. 0.0% packet loss and 0.8msec jitter.
>
>
> --
> Jeff Liebermann jeffl@cruzio.com
> 150 Felker St #D http://www.LearnByDestroying.com
> Santa Cruz CA 95060 http://802.11junk.com
> Skype: JeffLiebermann AE6KS 831-336-2558

--
Regards

stephen_hope@xyzworld.com - replace xyz with ntl



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