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We've been using G729 but the call quality was terrible. I swapped over to G711a and everything was perfect. We've got about 0.9mbit upstream. What's the maximum theoretical number of calls, and how would you calculate this. If you can suggest any ways of improving bandwidth use without compromising call quality, please go ahead too.

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G.711 runs at just under 100kbps per call (that's a 64kbps payload plus overhead).

G.729 shouldn't sound garbled - you should just get a slightly more "muffled" voice quality, and it does help with sites with limited bandwidth. However, G.729 also doesn't cope as well with congestion on access circuits (each packet contains more information, so packet loss is more significant)*

There's an excellent bandwidth calculator at

If you do have a congested circuit, then your options are limited - you would likely benefit from QoS on the circuit to prioritise your voice packets, but you'll probably want that at both ends, not just yours. Also, if there's aggregation of bandwidth going on between your site and the ISP (e.g. ADSL in the UK) then QoS is of limited value as the network may not support it.

  • You can get around this by modifying the packetisation for the codec, but it's not recommended and you do need both sides to know what they're doing in this case.
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I think the answer is going to be just a handful.

You should be able to test this pretty easily. Get an iftop or some other traffic watcher going on your asterisk and make a couple calls. Even just leaving the a couple phones or software lines open for a minute should give you an accurate bandwidth number since with G711 there isn't compression going on. As long as you don't have silence detection, what you see is what you get.

You'll see soon enough just how fast <1M fills up :)

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What would you recommend? We tried G729 but it sounded garbled... – chrism2671 Apr 15 '11 at 8:48
G.726 is a good intermediary between G.729 and G.711. If your equipment allows, you can also modify the sampling rate, which can tweak your bandwidth utilization for better efficiency. Sending fewer packets per second puts more payload in each one. This reduces the overhead costs of the call, but will make loss pronounced as more of the sound is in each packet. – mcmeel May 7 '11 at 5:27

64 kbit/s for each g711 channel. g729 can give an acceptable quality but

  • if you are using a pbx in your office you need to be sure that there is no transcoding between g729 and other formats (transcoding requires additional hw resources)
  • you need to set up QoS on your router/firewall and give max priority to the rtp stream between you and your operator

you can also try GSM, 13 kbit/s

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