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I have 25 Mbps on the downstream and 2 Mbps (250 KBps) on the upstream. I imagine upload speed is most important.

For a very high quality TeamSpeak audio codec, bandwidth transfer for a single user talking is approximately 7.5 KBps.

Therefore, if everyone was talking, I would be able to host (250 / 7.5) = 33 people without it lagging at all, correct?

However, there will never really be all 33 people talking at once, and even then, it would just lag slightly, I believe.

I was wondering if I am correct in all of this, how much my computer or network may lag with so many users, and how many users I would be able to approximately serve considering not all people talk at once.

Thanks.

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Your math works out correctly for the bandwidth calculation, but you also mention you want very high quality audio. VOIP is all about latency and jitter. It's simply not going to be possible to guarantee very high quality with a home internet connection where you don't have the ability to control upstream buffer sizes or perform QoS across the entire path of the conversation. –  fcorrao Mar 30 '13 at 21:50
    
@fcorrao: Perhaps I should have mentioned my network specs because it was recently upgraded. I have a ASUS RT-N66R Router. It's a pretty new model. Do you know what is required to perform what you mentioned in the latter portion of your post? –  John Seers Mar 30 '13 at 22:00
    
What @fcorrao is correct, latency and jitter are key factors in VOIP communication. To say that running a VOIP server on a connection is never going to work smoothly is incorrect. You're math is correct, you should be fine. I've been running a Ventrilo server on a home connection for years now. –  Rain Mar 30 '13 at 22:14

1 Answer 1

Yes, you will be constrained by your upload speed. Your calculations don't take into account packet overhead, for which you will need to add about 10kbit of bandwidth, meaning about 80k per call (depending on a number of things, like number of samples per second, compression) - so a total of 25 calls is probably reasonable.

I do not know much about Teamspeak (but I am familiar with VOIP in general). I believe that Teamspeak is not a codec, rather an encapsulation like SIP. If this is the case, I wonder if the codec you are using is ALAW or ULAW. This is relevant because those codecs provide "phone call quality" voice with very little processing overhead, meaning your server can handle a lot of them. If that is the case, you could look at a Speex based codec which will give you even better quality at a lower bandwidth utilization - but with higher CPU utilization.

This link might also be usefull to you.

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