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A Quick Introduction to VoIP

Packet data thrives over a voice network, but voice trembles at just the thought of traveling over a packet data network. VoIP? It just won't work!

Actually, it will work, and quite nicely -- if everything works just right. The trick is to keep latency, jitter and loss within limits. While that can be tough in a network that is built around applications that can tolerate all three, it's possible to pull a trick or two within the network, and at the endpoints, to make things work out.

The real key to sending voice over any packet data network is compression, and VoIP is no exception. Compression offers several advantages, one of which is the reduction of raw bandwidth required to support the information transfer. If we follow the trail of a voice signal over a typical digital network, we will see that it begins life as an analog signal, which is converted by a codec (coder/decoder) into a PCM format.

The result is packed inside an IP packet, along with a header for purposes of multiplexing, header error control, and identification of the application by port number. RTP (Real-time Transport Protocol) is run for end-to-end delivery services such as payload type identification, packet sequence numbering, time-stamping, and delivery monitoring.

On the receiving end, the process is reversed, and all is well -- at least as far as compression and decompression are concerned. The complication, of course, comes from the fact that the packets are not delivered by the network at the same pace that they entered it. Additionally, some packets may be lost in transit.

Remember that an IP network is a highly-shared packet network characterized by unpredictable levels of congestion. Latency is guaranteed, as is variability in delay (i.e., jitter). Loss isn't guaranteed, but it's highly likely, over time.

VoIP adjusts to latency, jitter and loss through various intelligent continuity algorithms employed by the receiving codec. These are designed to fill in the voids by stretching the voice frames received earlier and blending them with those received later.

This logic is embedded in predictive decompression algorithms which take advantage of the 10ms delay built into the compression/decompression processes to make the necessary predictions and do their stretching and blending. VoIP also makes use of various techniques for echo cancellation, as echo becomes perceptible when delay exceeds 15ms-20ms.

The actual end-to-end process is a little more involved, of course. In an enterprise-wide VoIP application, both the calling and called parties sit behind a PBX. The caller in San Francisco, for example, picks up the phone and dials the extension number of a co-worker in New York. The PBX checks its options for routing the call, courtesy of its LCR (Least Cost Routing) software.

Over a special link, the PBX hands the call off to a VoIP service provider. If the VoIP gateway has the sense that the call can be supported with acceptable quality of service, the call is accepted, and the above scenario plays out. The gateway compresses the voice data, packs it into a packet every 10ms, and off we go.

If network congestion levels remain low during the course of the call, conversation quality remains pretty good, but never quite as good as it is over the good old PSTN. If network congestion levels increase, so do latency and jitter, and packet loss may result. Voice quality suffers, and fond memories of the PSTN haunt the balance of the conversation.

If, on the other hand, the ingress gateway has the sense that current congestion levels are such that the quality of the call is likely to be compromised, the call may be routed over the conventional PSTN. Not all service providers, of course, offer the PSTN backup option, but many do in order to support business-class users.

So, voice data can be compressed in order to use shared bandwidth more efficiently; this can be done with little loss in voice quality, if everything goes just right.

Further, the decompression process can be sophisticated enough to smooth out some of the problems associated with latency, jitter, and loss of voice data over a packet data network -- within limits.

At this point, you have to ask yourself why in the world you would go to all of this trouble to run voice over a packet data network when the end result is uncertain quality that will never be as good as the PSTN, and which can be terrible when the network suffers congestion.

There are several answers, one of which is cost. VoIP is very inexpensive, at least in comparison to the PSTN. However, you still have to wonder if VoIP is worth the trouble in the face of PSTN voice calling at rates in the range of $0.04-$0.06 per minute. At rates perhaps as low as $0.02-$0.04 per minute, however, VoIP looks pretty good to the typical business enterprise. At rates in the general range of $0.04-$0.08 per minute, it looks real good to the small business and consumer market.

The cost issue takes on real significance in a multinational enterprise. Calls to Japan or South Africa, for example, may well be in the range $0.50-$0.60 per minute. VoIP starts to look real good at these prices. By the way, the way I see it is that VoIP will be huge, and for several reasons:

First, it will be inexpensive, even though quality will suffer a bit.

Second, many quality issues can be overcome by ever faster switches and routers, and ever faster transmission systems. (Admittedly, this is something of a brute force attack on a congestion issue, but bandwidth is pretty cheap these days.)

Third, VoIP offers tremendous advantages in terms of the fact that voice and data can be integrated over the same IP-based network, and through the same terminals in the form of integrated voice/data client workstations.

If you have any doubt about the integrated workstation thing, note that Microsoft announced VoIP support in its recent release of Windows XP. Imagine being able to exercise complete call control through your PC, share files with colleagues while discussing the contents in real-time, and in a truly collaborative mode over the same network at very low cost.

Imagine being able to negotiate with a call center agent to purchase an airline ticket or an automobile or a piece of furniture or clothing, all while looking at the same information on the Website, and all over the Internet, or other IP-based network. I repeat that this is going to be huge. There is very little doubt about that.

 

 

 

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