This depends on the codec being used. You can find details about the different codecs on the following website: http://www.voip-info.org/wiki-Codecs. Bandwidth also depends on your business model. For normal SIP call control bandwidth requirements are minimal, as you only need to allocate bandwidth for calls which will go through the RTP proxy (SIP-SIP calls only, since for SIP->PSTN and PSTN->SIP calls the RTP stream goes directly between UA and the gateway). How many calls you plan to send through the RTP proxy depends on the type of service provided and the planned number of customers.

Q: We are looking for 500 to 1000 customers for the first six months. Since we need bandwidth for SIP to SIP do you think 2 Mbps is enough, or do we need more? Since it all depends on the codec and both parties are on one network and running the same gateway, what do you think is needed?

A: This mostly depends on the number of simultaneous calls and codecs being used. 2Mbps will be sufficient for about 25 simultaneous calls using the g711 codec, 85 simultaneous calls using the g729a codec, or 120 simultaneous calls using the g723 codec. (See details at http://www.erlang.com/bandwidth.html) In my opinion, even 85 simultaneous calls are plenty for the number of customers you plan to have.

Q: If you need to link your network to another SIP service provider to exchange calls between two networks, since it will be SIP to SIP do you still need a high bandwidth or not?

A: If the customer has an Internet connection from his ISP (DSL, cable internet), and makes a call which is routed to your termination partner, the voice stream will go directly from the customer to your partner’s gateway, so only a few small UDP packets will be transferred to your network (call control).