VoIP Cookbook: Asterisk Softswitch
One of the best IP PBX Open Source SIP Proxy software available in the internet is Asterisk, which has so many features that allows us to establish our own VoIP infrastructure. Some of these features are highly useful for telecommunication operators, making Asterisk suitable for many applications ranging from small-to-medium-scale IP PBX to IP PBX having hundreds of telephone extensions. However we will not list these features right now, as you will find what these features are as you read along the rest of the CHAPTER, in the syntax code we explain later.
However, for high performance softswitch, it seems OpenSIPS seems to be better in handling high traffic request.
Such Asterisk scalability is possible because of Asterisk function called Trunking, which integrates various VoIP equipments, protocols, cellular, PSTN and even SIP provider to a same network. The number of trunks, theoretically, depends on the amount of available bandwidth and the speed of the processing machine that runs Asterisk.
So what do we need so that we can use Asterisk? The answer to this question depends on how many telephone extensions you want to have in your system and, importantly, how many concurrent calls Asterisk can facilitate. Ideally, the Specifically, you have to understand the following parameters:
- The number of outbound connections and their type (Analog, ISDM, T1, VoIP).
- The number of internal and external concurrent calls (the ratio between calls).
- The type of phone that will be used (Analog, SIP, H.323, MGCP).
- The type of codec that will be used.
- Whether transcoding process will be necessary.
- How reliable the system is.
- How many Asterisk machine that will be placed.
- The condition of your computer network in terms of processing speed, Quality of Service (QoS), * VLAN, and Power over Ethernet.
In general, a faster processor and the bigger the RAM, the more concurrent calls the server can facilitate. Since Asterisk seems to theoretically require around 30 MHz of CPU resources for every active channel, a 266-MHz CPU, for example, should ideally be able to facilitate about 8 concurrent calls, with the assumption that the Codec being used is G.711. Of course, in order to become an operator, you need to have a much more sophisticated server with faster CPU and higher RAM. But in order to understand what you really need, you can look into a variety of examples of hardware configurations and their maximal capability, which are available at http://www.voip-info.org. Through this site, you will also find the scripts required to simulate a call and put some load on the system.
Based on these considerations, you will know how much money you really need to spend. Spend sometime browsing the internet to make some comparison on internet telephony equipments and how much they cost. However, manufacturers, normally, do not show the price of the items they sell in their site. These price tags are usually shown in sites selling internet telephony equipments, some of them are:
- Digiumcards - http://www.digiumcards.com/
- VoIP on solutions - http://www.voipon.co.uk/
- The VoIP Connection - http://www.thevoipconnection.com/
The prices may vary, ranging from US$15 to US$50 per FXO or FXS. Meanwhile, IP Phone each cost between US$50 to US$150. You will of course get for less when you purchase them in large quantities. The cheapest you can get are the equipments produced in Taiwan or China. Some of them are LevelOne and Nexus.