|
These days, if you're starting a home or small office, chances are you might be considering one of the VoIP providers rather than getting another analogue phone line put in. However VoIP call quality and reliability depends on a number of factors. Here are some tips to improving your VoIP call quality.
There's a glossary at the end of this article.
Choose the Right Handset
Whether you choose to use an IP handset or an analogue telephone adapter, make sure you choose one with a proven track record. There are so many manufacturers, and many new ones entering the market each month, that it can be a difficult task to weed out the chaff. Many newer manufacturers ship their phones with pre-release firmware that can result in all sorts of issues, ranging from configuration difficulties through to dropped calls and choppy audio. A good community site for checking user reviews of handsets and ATAs is www.voip-info.org. This community site presents a great deal of information about VoIP in general.
Choose the Right Codec
A codec (shortened version of enCOder/DECoder) is the software algorithm that is used to convert your analogue audio speech into an (often compressed) digital form. There is a range of codecs available for VoIP. Most handsets and ATAs support at least two.
Generally speaking, the trade-off in choosing a codec is between bandwidth and CPU utilisation. A codec that requires a smaller bandwidth usually taxes the CPU more. Choosing what codec to use depends mostly on whether you are connecting to a PABX on your local LAN (usually the case in larger offices), or, as is more often the case with small or home offices, you are connecting to a "Virtual PABX" or broadband phone provider on the Internet. If you're connecting to your local LAN there are no bandwidth constraints so you should choose a full-rate codec such as G711. If you connect over the Internet however, you most certainly require some level of compression. In this case, a codec such as G729 or iLBC can give good results.
Because the Internet is a public, uncontrolled network, data travelling across it often takes a circuitous, unordered route. Packets of data often arrive out of order, and with varying levels of delay (in VoIP jargon, this is referred to as jitter). Some codecs deal with jitter better than others; for example, G729 is particularly resilient to jitter. In some cases, packets are lost altogether. Virtually all VoIP codecs react badly to packet loss.
When using a high-compression codec, which necessarily taxes the CPU more, it's important to ensure you have a robust phone with the latest firmware installed (see note above regarding handset selection). Poorly written firmware can often result in choppy audio when using a high compression level because the CPU simply can't keep up with the requirements of the algorithm.
Prioritise Your Internet Traffic
Selecting a good handset and the most appropriate codec is no use if your VoIP calls are having to compete with other Internet traffic on your broadband connection. You could be having a perfectly crystal clear phone call with a client one moment which is suddenly interrupted with all sorts of drop-outs and distortion because your computer is busily downloading several MB of email.
Most modern SOHO routers have a facility to prioritise various kinds of traffic competing for use of the connection. The simplest form enables you to give a higher priority to traffic destined for a particular IP address and port number or range. You should set this to be the IP address and audio UDP port range for your VoIP provider (they will be able to provide you with this information). Another simple form of prioritisation is by internal IP address - however this requires that your phone handset have a static IP address (as opposed to being automatically assigned one by your router).
Some modern routers are smart enough to recognise VoIP traffic, and automatically prioritise it over other kinds of traffic. Indeed, some manufacturers are now coming out with inexpensive devices that connect between your router and your broadband modem. These devices require no configuration and will automatically prioritise voice traffic.
Yet other routers provide what is called QoS (Quality of Service). These are special tags that can be attached to IP packets that are read by some ISP's routers and used to prioritise traffic within the ISP's network (once it has left your broadband modem). Most ISPs that also offer broadband phone services support some sort of QoS. It's best to consult with the ISPs technical department about exactly how to set this up. While valuable, QoS is not going to help unless you have already sorted out your traffic priority internally using one of the above methods.
Glossary
- ATA - Analogue Telephone Adapter
- Codec - (enCOder/DECoder) Software algorithm for compressing audio
- Firmware - Software installed in the chips in an IP phone or ATA that handles the making and receiving of calls, including processing of audio with a codec
- ISP - Internet Service Provider
- IP - Internet Protocol
- PABX - Private Address Branch Exchange
- Router - A device which channels data from your home or office network to the Internet and back
- UDP - Universal Datagram Packet: the packet type used for transmitting audio over the Internet
- VoIP - Voice over Internet Protocol
Michael Pollitt is an experienced SOHO business operator, with several successful SOHO businesses, including ConferenceOne, a telephone conferencing service, and LB Broadcast Systems, a developer of closed captions software. |
|
|
|
