Network requeriments of VoIP ?

Hi,

Well, basically the title says it all. If I set a wireless network for data and VoIP what would the requirements be?

What are the bandwith requirements of each VoIP call ?

Whats is the maximum latency allowed for a VoIP call ?

Are there other requirements ?

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Reply to
landuchi
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24-96 kbps

200ms at absolute most.

A good router, a good wireless access point, and good engineering. Dont slap it together... spend the money to do it right the first time and you'll be happy with your setup.

Chris NetSteady

1-866-678-9434
Reply to
NetSteady

landuchi hath wroth:

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.

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.

Depends on the codec. See chart at:

That also varies with the codec. See:

for various calculators. Be sure to use half-duplex for wi-fi. This looks better:

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.

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).
Reply to
Jeff Liebermann

Jeff Liebermann hath wroth:

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:

There's some notes in there on jitter.

VoIP and Jitter tester:

I'm having a good day. 0.0% packet loss and 0.8msec jitter.

Reply to
Jeff Liebermann

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.

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.....

Reply to
stephen

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