My CISCO Journal

Different protocols consume different amount of bandwidth. In most of the cases the bandwidth consumed by VoIP codec is not that much, the extra burden is caused by different protocol headers. VoIP bandwidth calculation and planning is one of the most important part in designing a VoIP Network.

To make this task easy, i cam across a very interesting website. The website is:

http://www.bandcalc.com

This is an interesting site, which you can use to calculate the bandwidth required for a VoIP network using different parameters like, payload size, frames per second etc. Please explore and enjoy this interesting website.

What is an Erlang?

Erlang is a traffic measurement unit commonly used in telecom. It is used to describe a traffic volume of one hour.

Example: 20 calls in an hour with 5 minutes of conversation average.

You calculate the number of Erlangs as below:

Traffic minutes in the hour: 20 x 5 = 100 minutes
Hour of traffic inside one hour: 100/60 = 1.66 Erlangs

You can get these measures from a call logger and use it to design your network to calculate the number of lines required. Once the number of lines is known, it is possible to calculate the bandwidth requirements.

• G.711 has a Mean Opinion Score of 4.3-4.7 and uses 80 kpbs (if you send 50 packets/second with 20 ms of RTP payload per packet) or 74.7 kbps (@ 30 ms, meaning 33.3 packets/second).
• G.729 (NOT G.729a) has a MOS of 3.9-4.2 and uses 24 kbps @ 20 ms or 18.7 kbps @ 30 ms.
• G.729a has a MOS of 3.7-4.2 and uses 24 kbps @ 20 ms or 18.7 kbps @ 30 ms.
• G.723 has a MOS of 3.8-4.0 and uses 17.1 kbps @ 30 ms.

MOS is what nontechnical people think about each codec (5.0 is perfect). All of the above numbers are in EACH direction, so total bandwidth is double the above figures.

As you can see, there is very little quality or bandwidth difference betweeen G.729a and G.723. However, G.729a can send 50 packets/second, each packet containing 20 ms of voice payload. G.723's lowest setting is 30 ms. I think 20 ms sounds better than 30 ms because of smoothing (used to fill in for late packets).

It would be a very good idea to turn on silence suppression ('Silence Supp Enable' = 'yes'), because it will reduce your bandwidth usage by 65%. (Apparently each person in a two-person conversation only talks about 1/3 of the time.)

There is also the issue of transcoding. Whenever one lossy codec is decompressed and then transcoded into another lossy codec, voice quality always gets worse, so you want to avoid transcoding. I've read that G.729 is commonly used for overseas calls, so if your adapter compresses your voice using G.729 and then your long distance provider decompresses your voice and then recompresses it using G.729, no voice quality is lost. Transcoding is especially noticeable in cell-to/from-VoIP calls, so if your adapter supports GSM (SNOM does) and you choose it as your default codec, your voice quality will be better when calling GSM cell phones.

Finally, I've tried VoIP over dialup, and it's no fun, because of dial-up's high latency and jitter. It seems like there should be a Nextel-style "push to talk" adapter/software for high jitter VoIP connections (like satellite and 802.11 mesh networks), but I don't know of any. However, my test was before Sipura added the 'Network Jitter Level' = 'extremely high' setting. That setting might be enough to make a VoIP over dialup conversation workable if you're willing to take turns talking and say 'over' when you're done talking.

For more interesting discussion, please visit : http://www.dslreports.com/forum/remark,14760386