Implementing VoIP

Most VoIP implementations follow the ITU H.323 standard. This consists of four basic components: the terminals, IP gateways, gatekeepers, and Multipoint Control Units (MCUs). Out of these, the first two are key elements, while the other two are optional components.

The H.323 standard was originally a multimedia standard meant for transferring audio and video data over a network such as Ethernet or Token Ring, which doesn't provide guaranteed QoS (Quality of Service). QoS happens when you can guarantee the timely delivery of information on networks, control bandwidth, set priorities for selected traffic, and provide a good level of security. The H.323 standard consists of a set of audio and video compression algorithms and control protocols. We've covered the compression algorithms in a separate article, and we'll cover the protocols here. So let's start with the various components of a VoIP setup. 

There are different ways in which you can implement VoIP, all of which depend on the size of your network infrastructure, and the kind of application you'd like to use it for. Before implementation, you have to ensure that there's sufficient bandwidth on your network to handle the voice traffic. If not, you'll have to upgrade infrastructure to support the voice traffic. 

Setting up VoIP

A basic setup between two locations will consist of an IP gateway on either location. The gateway can be designed inside a PBX or a standalone device such as a router. On one side of this gateway will be the PSTN (Public Switched Telephone Network) or a company's IP WAN, while the other side would have the company's internal network. PSTN connectivity could be through wireless, ISDN, or analog telephone lines. The IP WAN link could be leased line, frame relay, ATM, etc. The gateway is essentially a device that translates protocols from the network onto the PSTN network. So, the one you choose will depend on the type of connectivity you're using, and the protocols you'll use, which in this case will be H.323. You can't, however, interface the gateway to a PBX that's connected to the PSTN yet, not because of technology issues, but because the law doesn't permit it; you can use it only to connect your internal network with your WAN. The gateway also has to interface with a router on your company's internal network, which will split the voice and data and assign priorities.

On the internal network, you need terminals that are capable of voice communication. These could be PCs with headphones and a microphone, or IP telephones. The IP telephone will hook up directly to your network through an RJ45 Ethernet port. Terminals support two-way communication over IP. They encode audio signals for transmission, decode them for reception, and also perform some control functions. A terminal uses certain protocols for communication. These include the H.245, which negotiates channel usage and capability. It also uses Q.931 for signaling and control; and RTP/ RTCP (Real-time Protocol/Real-time Transport Control Protocol) for sequencing the audio packets. These packets are sent via UDP (User Datagram Protocol). And finally, there's the RAS protocol, which sends Registration/Admission/Status signals to the gatekeeper. All control protocols are sent using TCP.

Other devices

In addition to gateways and terminals, there are devices called gatekeepers. Simply speaking, a gatekeeper acts as a routing device for voice calls. They do address translation to determine the calling terminals. They can also control access given to the other VoIP devices such as terminals, gateways, etc. They determine how much bandwidth is required for a voice call, and can perform other functions like call authorization, bandwidth management, call management, and directory control. Finally, there's a device called Multipoint Control Unit, or MCU. This is meant to provide conferencing facilities between three or more H.323 terminals or gateways. 

Anil Chopra