While bandwidth is now abundant and cheap, particularly on a local area network (LAN), moving what could potentially be large continuous amounts of data without interruption from one point to another within a given period of time may not be easy, especially over a wide area network (WAN) link. The average PSTN call runs at 64Kbps. That 64Kbps channel needs to be open and unaffected for the duration of the call. Naturally, not many VoIP installations could afford that type of sustained traffic on the network, particularly large deployments, therefore the dreaded technology C-word must be used: "compression".
Go on, compress me
Of course, with compression comes loss of audio quality, say the knockers. The most commonly standard used with VoIP is H.323, which incorporates the G.723 codec. This can take a 64Kbps stream of data and squash it down to a mere 5.5Kbps or so. Before you get too excited, you also need to take into account the overheads that it takes to transmit that data, and in some situations these could be quite high. For VoIP to work effectively over WAN links, there needs to be low jitter, low packet loss, a relatively high-speed connection between the endpoints, and less than 200ms delay.
Long pauses, unexpected dropouts, or any other strange phenomena not usually associated with land line telephone calls are unacceptable in a VoIP deployment. The service has to be as good as normal landline telephone services. Jitter buffers in the technology help to reduce the effect that jitter can have on the connection, but ultimately the connection is only as healthy as the network it is running over.
This takes us back to the compression protocols -- surely if something is removed through the compression then the quality can't be the same. Interestingly, where the most savings come in the G.723.1 standard are in the pauses between words. Believe it or not, up to 50 percent of a telephone conversation is silence. Please don't even mention music on hold; most vendors have some very interesting ways of dealing with it.
However, when the data is decompressed at the other end, if silence was inserted between the gaps it would sound odd because usually there is some background noise or even the usual reassuring line noise. Various developers deal with the situation in different ways. Some introduce a "generated" hiss or line noise, so that the user of the system does not think that the line has dropped every time the speaker pauses for breath. Another solution is to randomly sample some background noise from each end of the phone conversation link and inject that back into the blank gaps in the conversation.
What's out there?
We looked at VoIP systems from Avaya, Cisco, Nortel, and Zultys. Each vendor participating in this review was asked to provide either a demo site at their premises or a live site where their VoIP products were in operation. The Test Lab then visited these sites, spoke with the vendors and their engineers about the products being used as well as other products in the vendors range which could also be deployed.
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