Skype Gives Away SILK Codec - Has Wideband Telephony Finally Arrived?

jon-christensen.gifJonathan Christensen, General Manager, Audio & Video, Skype is keynoting at eComm and announced some major news today - namely that Skype is going to give away their signature SILK wideband audio codec (aka Super Duper Weenie Wideband Codec) in hopes it will help make wideband telephony adoption more prevalent. This could finally help make wideband telephony in the VoIP industry a reality. Rich Tehrani is a strong proponent of wideband audio, but I'm still a bit of a skeptic myself. I like wideband audio, I just don't think there is a 'rush' to adopt it. To get true wideband audio, you need wideband audio from end-to-end. Once you touch the PSTN (analog, T1/PRI), you're no longer using wideband audio.

Still, if IP phones (Polycom, Aastra, snom, etc.) start adding SILK to their phones and SIP trunking service providers support the codec, then you could see a rapid adoption of wideband audio.

Jonathan remarks on his blog:

Today marks a significant moment in the journey of Skype. Some of you may be aware of the SILK speech codec which is included as part of Skype 4.0 for Windows - it's what enables super-wideband audio and optimizes call quality, even in low network bandwidth environments. It transforms sound quality for Skype calls, and if you haven't already tried 4.0, believe me: you should.

I'm speaking at eComm today, and have some very special news to bring our developer community - we're making the SILK speech codec available for a royalty free license by third-party software and hardware developers soon.
What does this mean for the future of audio on the web?

SILK is Skype's signature super wideband audio codec which achieves super wideband audio quality using 50% less network bandwidth than previously required. It is the outcome of a three year long development process in the Skype labs, which focused on four things:

* improving audio bandwidth going from 8 kHz to 12 kHz, meaning that a SILK conversation sounds like you are in the same room as the person you are speaking with
* providing real-time bandwidth scalability to deal with degraded network conditions
* balancing codec optimization between voice, music and background noise, each of which can have an impact on the overall user experience
* delivering a robust solution that delivers a more consistent audio experience, regardless of network conditions and an individual user's voice signature

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