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A codec is software that converts audio signals into digital frames and vice versa. Codecs are characterized with different sampling rates and resolutions. Different codecs employ different compression methods, using different bandwidth and computational requirements.
Choosing the best codec for particular network conditions may considerably increase the quality of voice calls. If the network has low effective bandwidth, choosing otherwise great lossless G.711 codec would be a mistake, as the quality of the calls would suffer because of bandwidth limitations and lost packets rather than codec quality. If there is less than 64 Kbit/s of available bandwidth, picking a low-bitrate, high-compression G.729 or G.723 codec is much more appropriate.
Note that while a local area network (LAN) may provide high bandwidth, external calls may be subject to bandwidth bottleneck in the upstream. ADSL and cable network providers often offer limited upstream bandwidth, which results in upstream congestion if multiple VoIP calls are carried concurrently. In this case, low-bandwidth codecs may provide better results. G.711 (PCM), a high-bandwidth codec, provides the best audio quality yet consumes the most bandwidth. G.729a (CS-ACELP), G.723.1 (MP-MLQ) and G.726 (ADPCM) offer varying conversation quality, sorted by decreasing relative quality.
The codec choice will not take place automatically. A system administrator must specify and prioritize codecs available to the particular VoIP system. Taking care of the wrong choice of codec may significantly improve conversation quality. By logging and displaying the session flow, a network analyzer allows seeing the codec negation process, i.e. the codecs available to the endpoints, as well as the final negotiated codec choice.
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