In a call centre environment, what bandwidth would be required for 55 lines? This would be on SIP protocol, would we be best to use G729 codec as i know that our sip provider supports this? What type of internet connection would be best for such a task? Would SDSL 1:1 be sufficient (im in the UK)?
Hoping someone with experience in the field can give me some tips...
A good rule of thumb comes from old telephony .. an phone T1 line 24 voice channels and a control channel. This equates to roughly 64Kb/s per phone line. This rule works (roughly) for uncompressed VoIP. There are various codecs that do compression .. the G729 should roughly triple that, so get 72 voice channels on a T1.
If you expect all 55 phones to be on external calls at the same time, you will utilize roughly 75% of a T1. That is pushing the realistic limits. A single T1 will be fine if you expect less than 80% or so phone utilization. Generally speaking, as you push the bandwidth limits you will see degraded quality before you see calls fail, so a T1 should be OK for the occasional peak to 100%.
IMHO the kind of connection matters little, as long as you get low latency between between the phone and the "switch" (aka the VoIP server).
Were I in your shoes I would get a T1 and watch the performance, call quality, the number of phones, and the average phone utilization. I would also be careful about compression .. it can really do a number on call quality, particularly when combined with tight bandwidth.
Also, if you are going to put phones and PCs on the same network, make sure you get QoS running properly on the internal switches and the internet router. It would suck to have enough bandwidth but have poor call quality because of the call center reps watching videos on YouTube or opening files on the server. If you don't know how to deal with QoS, either get someone who does, or put the phones and computers on separate networks and use separate internet connections.
The best place to start would be with the voip equipment vendors documentation. Barring that, there are several web based calculators. Here are a few to start you off with:
I'm no expert at SIP, but here is what I've found out earlier:
Number of connections * bandwith (kbit/s) per connection = bandwith for SIP
You'd also want to reserve some data for other traffic - say 20% extra.
And - for the love of every person who's going to make calls - Use QoS agressively if you're using the link for anything other than SIP. This also applies to your internal routers/switches that the VoIP traffic runs through.
If this system is an important part of your business and you need high up-time I'd strongly advise against SDSL, it just can deliver the quality you'd need, it's just not designed for that. I was called in to sort a very similar problem with a 100-user call centre a few years ago just outside London, we moved them to a pair of 10Mbps BTnet leased ethernet lines and the problems literally disappeared overnight.
We are looking at VoIP right now, and our vendor said that each active SIP channel would use 64kbps. For 55 active calls, you would need about 3.5mbps sustained bandwidth. In our case, we'll never have more than a couple calls running simultaneously, so 200 kbps would be plenty.
As noted by joeqwerty, you should check actual bandwidth consumption using some available calculator like the one hosted at http://www.erlang.com/calculator/lipb/
As a rule of thumb, we usualy reserve 35kbps symmetrical for each active g729 call, so 55 simultaneusly used lines would requiere 2 Mbps just for VoIP. If the link will also be used with some other app, QoS and traffic policing is a must.
Also, with VoIP, jitter and latency is as important as bandwidth, so you should go with a good quality link.
For 55 g729 calls, you will use a lot of CPU for compression. Don't discard using some hardware assisted coding, Digium has a card just for that. You will spend on the card but save on the codec.
