by Philip Carden

Regardless of what’s in between, a phone conversation between two people requires that each has both a microphone and a speaker. In the traditional telephone, the microphone is located in the mouthpiece and the speaker is located in the earpiece. In an analog telephone (like the one you have at home), the voice signal produced by the mouthpiece is sent directly along the wire to a telephone exchange or a local PBX.

If you’re going to use IP telephony, you’ll still need a microphone and speaker. Those could be the microphone and speaker supplied with your PC or built into a PC-attached headset. But they could equally well be provided by a traditional analog telephone attached to an IP telephony enabled PBX (a so-called iPBX) or by a telephone plugged into a data port that supports IP telephony directly (an IP telephone). Regardless of whether it’s a PC or a traditional telephone attached to an iPBX or an IP telephone, the basic mechanics of how an IP telephone call works are the same.

The call

So what happens when you want to make a call? First of all, after you’ve dialed a telephone number or clicked on a name, signaling is required to determine the status of the called party — available or busy -- and to establish the call (signaling is used for many other things too, but more on that later). Next, when the conversation starts, the analog signal produced by the microphone needs to be encoded in a digital format suitable for transmission across an IP network. The IP network itself must then ensure that your real-time conversation is transported across the available media in a manner that produces acceptable voice quality. Finally, it may be necessary for the IP telephony stream to be converted by a gateway to another format — either for interoperation with a different IP based multimedia scheme or because you are placing a call to the traditional public telephone network. The overall technology requirements of an IP telephony solution can therefore be split into four categories — signaling, encoding, transport and gateway control.

The standards

For each of these areas there exist standards. Unfortunately, in the key area of signaling there are two competing sets of standards — H.323 (an ITU standard) and SIP (Session Initiation Protocol, an IETF standard). This is why discussions of IP telephony often seem to boil down to an H.323 vs. SIP argument. It is important to remember, however, that neither H.323 nor SIP alone make up a complete set of IP telephony protocols — they are just the competing standards for signaling. Until recently there was a similar situation with gateway control protocols, with competition between IPDC (IP Device Control) and SGCP (Simple Gateway Control Protocol). Now, however, the industry appears to have agreed on a third standard, called MGCP (Media Gateway Control Protocol), which combines the advantages of its two predecessors. For details on which standards relate to signaling, encoding, transport and gateway control, see the protocol tables below.