Uncoupling low-bit-rate voice and VoIP
 
|
|||
 | |||
|
Sign up to receive this and other networking newsletters in your inbox.
In the last newsletter we disconnected voice over IP and voice over the Internet. In this one, we're going to disconnect the necessity of using a low bit-rate coding algorithm for voice with VoIP.
This close association between using a very efficient voice coding algorithm and packet voice in general evolved naturally. Five or so years ago, the major rationale for using voice over frame relay or VoIP was toll bypass, especially for international telephony. In order to pack as many voice conversations as possible onto a single frame relay circuit, the relatively low bit rate 8K bit/sec G.729A algorithm was chosen for voice over frame relay.
The even lower bit rate G.723.1 algorithm at 5.3K bit/sec and 6.3K bit/sec was chosen as the default for VoIP. Like the frame relay algorithms, this enabled the packing of many calls onto a single connection between sites. It also was designed so that VoIP could be used by consumers for Internet telephony over relatively slow dial-up connections.
The world has changed, and so should our thinking. There's no longer a need to maintain a mandatory association between VoIP and low-bit-rate voice. The algorithms sound great to almost everybody's ears. The link below gives some samples of coded voice using the various algorithms.
But whenever we start by proving that the coding for VoIP sounds as good as traditional 64K bit/sec pulse code modulation (PCM) voice, we open the door for disagreement. And that's a door that could be left closed.
Essentially all equipment today allows you to choose your preferred algorithms. If you like the low-bit-rate sound, then go for it. But, just as we pointed out last time that equating VoIP with voice over the Internet could be a showstopper, this shouldn't become a major stumbling block. If you prefer the sound of 64K bit/sec PCM, then use PCM. In fact, even if you don't prefer PCM but you just don't want to have to answer questions about the quality of the voice coding algorithm, the use PCM.
Next time we'll dig a little deeper on this topic.
RELATED LINKS
Steve Taylor is President of Distributed Networking Associates and Publisher/Editor-in-Chief of Webtorials.Com. For more detailed information on most of the topics discussed in this newsletter, connect to Webtorials.Com, the first Web site dedicated exclusively to market studies and technology tutorials in the Broadband Packet areas of Frame Relay, ATM, and IP.
Larry Hettick is an independent consultant, with 19 years of experience in telecommunications and data communications marketing and product management for service providers and equipment vendors. He can be reached at larry@larryhettick.com
You can reach the authors at taylor@webtorials.com or larry@larryhettick.com.
Convergence archive
Past newsletters.
