PREVIOUSLY: VoIP codec basics
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When running VoIP you have a choice of several codecs, mostly based on the International Telecommunication Union's ITU G.7xx protocol. The resulting call quality is measured using a Mean Opinion Score (MOS) of between one and five. Listeners rate the audio quality using test sentences, with a score of five meaning there are no perceptible call quality issues.
Of the ITU G.7xx codecs the best call quality is achieved using;
- ITU G.711: 64 kbps
This codec uses a narrowband 8 kHz sampling frequency and employs Pulse Code Modulation (PCM) to sample the audio at regular intervals - boosting the quality but also the bandwidth required. G.711 employs the Âµ-law algorithm (in North America & Japan) or A-law algorithm (Europe and the rest of the world). The Âµ-law and A-law algorithms compress 14-bit and 13-bit PCM audio samples down to 8-bit. Calls using G.711 generally sound better than PSTN and the equivalent to ISDN, but they consume a lot of bandwidth. With overheads, G.711 requires at least 85 kbps of bandwidth per phone call.
- ITU G.722: 48, 56 or 64 kbps
Uses wideband 16 kHz sampling frequency but employs Sub-Band Adaptive Differential Pulse Code Modulation (SBADPCM) to encode the 14-bit PCM values as differences between the current and the previous value (which are then compressed to 8-bit). It can also vary how often it samples the audio. By combining these tricks, SBADPCM can reduce the number of bits required per sample by more than 25 per cent compared to PCM used in G.711.
- G.722.1 uses a lower bit rate to consume 24 or 32 kbps
Also known as Siren7, it's based on Polycom's SIREN codec. G.722.1C, or Siren14, consumes 24, 32, or 48 kbps. Another variation, G.722.2, also known as AMR-WB (Adaptive Multi-rate Wideband), utilises even greater compression to consume 6.6 to 23.85 kbps. It also has the ability to vary the bit rate according to network conditions.
- Speex: 2 to 42 kbps
This codec supports ultra-wideband 32 kHz, wideband 16 kHz and narrowband 8 kHz sampling frequencies and is compressed using Code Excited Linear Prediction - an algorithm designed specifically for low bit rate speech. It can also vary the bit rate. Although Speex has no licensing fees, it is more likely to be found running on open source VoIP systems, such as Asterix, rather than built into proprietary VoIP equipment.