On August 5, 2008, Polycom announced the royalty-free availability of its wideband Siren 7 codec (ITU-T G.722.1). I reached out to Jeff Rodman, Chief Technical Officer and Co-Founder of Polycom to get some more insight into what the Siren 7 codec represents and what it means to the market that Polycom is offering the codec royalty free.

 

 

JR: Conventional telephone-quality audio is known as “narrowband” because it carries only about one-fifth of the range of the human voice. This elimination of 80 percent of the voice is often the biggest reason that people are so hard to identify and understand on a phone call. With IP telephony, it becomes much easier to send HD Voice (also called “wideband”), which carries most or all of the voice, and this dramatically increases intelligibility and decreases fatigue. A “wideband codec” is a standard algorithm, a piece of software that compresses and decompresses wideband speech to more easily fit over an IP network.

 

JR: Siren 7, more formally known as ITU-T G.722.1, is a wideband audio codec that delivers 7 kHz audio, which is the most important part of human speech, but uses only half the data rate of the older G.722. G.722 is the “grandfather” of 7 kHz wideband VoIP codecs, and the most widely deployed so far. But while G.722 operates at a data rate of 64 kbps, Siren 7/G.722.1 can deliver equivalent quality at half this rate, and even operates well down to 24 kbps. One difference between these two is that the older codec uses ADPCM compression, while Siren 7/G.722.1 is a transform-based algorithm that uses a set of sophisticated frequency analysis techniques to achieve more efficient coding. In spite of this improved performance, the more modern Siren 7/G.722.1 uses only about one-half of the processing required by G.722.

 

JR: Polycom has a long tradition of supporting open standards in the industry. By now making this highly efficient standard available on a royalty-free basis, we expect to see accelerated adoption of wideband audio in the telephone industry, to the benefit of every telephone user as well as those of us in the industry itself. HD Voice provides far more vibrant, lifelike conversations with more than twice the clarity of narrowband audio, and this can now become available to any user.

 

JR: With the rapid transformation of business telephony from circuit-switched to VoIP — more than 70 percent of new lines are VoIP — there is an accompanying move from low-fidelity 3 kHz audio-only telephones to devices that take advantage of the IP network to deliver integrated higher-fidelity sound, features, applications, and the whole range of integrated communication capabilities sometimes known as “Unified Communications (News - Alert).” All of this is happening because with the move from plain old telephone service (POTS) to VoIP, the network is suddenly transparent and can deliver an enormous new range of capabilities.

 

The transition from narrowband audio, or 3 kHz, to VoIP wideband telephony is already accelerating. One element enabling this transition is the addition of wideband-capable codecs, from 7 kHz to 20 kHz, to the codec repertoire, and this is why Polycom’s liberalization of the Siren 7 codec is so significant.

 

There are two trends to keep in mind in VoIP audio bandwidth today; one is strategic, and one is technical.

 

The strategic trend is this: VoIP telephony is moving toward full bandwidth 20 kHz sound, because the VoIP endpoint is undergoing transformation to a multi-purpose, multimedia device that integrates communications, applications, and even entertainment. As you have seen, there’s little cost or bit rate penalty in going to wideband telephony using modern codecs — even the full-band G.719 codec has a lower bit rate than G.711 — and competitive pressure will drive VoIP vendors to achieve full human compatibility in a very few years. Some applications will remain at the voice-friendly 7 kHz point due to tight cost or size constraints, but we’ll see an increasing tide of fully capable 20 kHz VoIP systems with unified capabilities.

 

The technical trend follows from the strategic: which codecs will bring us to this 20 kHz world? At 7 kHz, G.722 has been mature, free, and widely deployed in endpoints and in PBXs and softswitches, but it is now joined by royalty-free G.722.1 at half the bit rate and half the processing load. Another codec, G.722.2, is deployed in applications where its higher cost is offset by its very low bit rate and high quality, much of this driven by cellphones. Its adoption there will push the network, and consequently wired endpoints, to follow.

 

These three codecs form a functionally complete set for 7 kHz performance. The choice at 14 kHz is G.722.1 Annex C because of its maturity, modest bit rate and processing needs, and zero-cost license (also provided by Polycom). And finally, 20 kHz performance in the VoIP world will come from the ITU’s new G.719.

 

JR: As an approved ITU standard, Siren 7/G.722.1 is a final specification, and implementations that comply with this standard are thus fully interoperable. Many companies have been shipping products using this codec for years, and we expect to see its increasing adoption in the IP telephony industry due to its low data rate, high quality, low cost, and low processing load. Polycom also offers its 14 kHz sibling, G.722.1 Annex C (Siren 14) on a royalty-free basis, and has successfully completed collaborating on a 20 kHz codec with Ericsson (News - Alert), which received final approval as ITU-T G.719.