Voice over IP: Security, stability, success

Summary:If you're thinking about voice over IP -- and sooner or later you'll have to -- we take a look at the steps involved in getting it set up and what's on offer from four major vendors.

Bandwidth issues

  Voice over IP

 VoIP at a glance
 Getting started with VoIP
 Bandwidth issues

Cisco systems

 Final words
 Glossary of terms
 About RMIT
One of the main criticisms of VoIP is the amount of bandwidth required for each conversation, and the knockers are sometimes quite justified to pick on this point. Not only must there be adequate bandwidth, the quality of the link must be maintained throughout each call to ensure the users are not affected. At the end of the day if a user is transferring a 3GB file from a workstation to a server, they will not really notice if it takes eight seconds longer one time than the last time due to network traffic impacting on the transfer time; it is all transparent and handled in the very nature of the TCP transmissions. However the same problem with a VoIP call would definitely be noticeable and unacceptable to the participants of that call because it is real-time. And real-time is the key. While VoIP may use TCP packets to set up and establish the call parameters, it mostly uses UDP packets to send/receive the voice traffic. Ensure that your firewall supports forwarding of UDP traffic. The two issues here are bandwidth and quality of service (QOS).

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.

Topics: Unified Comms, Networking, Reviews

Kick off your day with ZDNet's daily email newsletter. It's the freshest tech news and opinion, served hot. Get it.

Related Stories

The best of ZDNet, delivered

You have been successfully signed up. To sign up for more newsletters or to manage your account, visit the Newsletter Subscription Center.
Subscription failed.