The VoIP Mag

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Enterprises have traditionally maintained separate networks for their voice and data traffic. Their circuit-switched voice networks provide the controlled environment needed for high-quality voice conversations while packet-switched IP networks deliver the flexibility and low-cost bandwidth needed to support ever-changing data requirements.
But the days of separate voice and data networks are numbered. Driven by new voice over IP (VoIP) technologies and the need to reduce network costs, many enterprises are designing new converged networks capable of handling both voice and data. The key challenge is to reconcile the performance requirements of voice with the unpredictable nature of data on a single network.
This white paper analyzes VoIP performance and bandwidth requirements and shows how Expand Networks ACCELERATORs help to deliver the required performance while reducing WAN costs in converged networks.
The VoIP Performance Challenge
The motivation for running voice over IP networks is to eliminate the expense of maintaining separate voice and data networks. It sounds easy enough to run voice over IP network – just encapsulate digitized voice in IP packets and go. Digitizing and packetizing voice is fairly straightforward, but there’s one other key issue that is much tougher to deal with.
The key challenge in building converged networks is performance. Voice communications has much more stringent performance requirements than data communications. The best way to understand voice performance requirements is to analyze the traditional voice communications network – the public switched telephone network (PSTN).
First, with the exception of most “last mile” copper loops, the PSTN is a digital network. Since human voice is analog, voice traffic must be digitized before it enters the network and then converted back to analog on the receiving end. Pairs of codecs (coder/decoders) at the endpoints perform the conversions between analog and digital signals.
To provide high quality voice the codecs use a technique called Pulse Code Modulation (PCM) that samples analog voice every 125 microseconds (1/8,000 of a second) and digitally encodes each sample as an 8-bit code. Since 8,000 of these 8-bit samples must be transmitted every second, PCM requires 64 kbps of bandwidth for each call.
To ensure the quality of each call, the PSTN uses multiplexing and circuit-switching technology to allocate a fixed 64 kbps channel for the duration of each call. Since the required bandwidth is always available, there is very little end-to-end latency, no jitter (variation in latency), and virtually no data loss. The net result is a consistently high level of voice quality, called toll quality.