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HDVoice & Asterisk: Hearing The Siren’s Song – The Finale

Asterisk & HDVoiceHaving read & listened this far into this series you should now have some grasp of how narrowband (G.711) compares to wideband (G.722/G.722.1) and even super-wideband (G.722.1C) audio for telephony applications. The differences in many cases are quite pronounced, even startling. What you hear in the examples are just the most obvious properties of the encoding, sampling rate and by implication, the available audio bandwidth. It’s worth understanding a bit more about the evolution of the role of the codec over time. This will help you frame up how the Siren codecs fit into the Asterisk realm.

Back in the early days of the transition to digital in the PSTN the entire network architecture was TDM-based. This is the technological foundation that gives us the codec most commonly deployed, namely G.711. On a TDM network when a call is placed a pathway is created that provides absolutely assured bandwidth to the call. Since the network is such a known quantity the historical role of the codec was very simple. True to the roots of the term “codec” it merely encodes (or decodes) the raw digitized audio to/from a digital stream in a standard format and at a constant bit-rate. The codec is blissfully unaware of its operating environment.

There are all kinds of other processes that may be performed on the audio streams. There are source audio issues of gain control, echo cancellation and background noise suppression, then network issues like jitter buffering and packet loss concealment, just to name a few. All of these additional processes, when they occurred, were done outside of the codec. They were traditionally not part of the codec itself.

Flash forward about forty years to 2008 and the introduction of Skype’s SILK codec, the newest development on the codec landscape. Things are now very different. Over time the role of the codec has grown to include various of the processes that were once pre- or post-processing. Whereas bandwidth was once a constant, the new-found dominance of IP networks means that bandwidth is always a variable.

The current state-of-the-art in codecs, SILK is adaptive. That is, it’s aware of it’s operating environment. It changes its operation to best leverage the network conditions of the moment. It can alter its sampling frequency as well as qualitative factors in the compression process, ensuring that it gets the best possible call quality out of the available bandwidth. SILK can sound worse than cell phone or almost like a CD. It does all of this this dynamically in response to instantaneous network conditions.

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