With the release of Skype 4.0 for Windows beta 3 a year ago February, Skype first introduced their superwideband SILK codec. A month later at eComm 2009, Jonathan Christensen announced SILK’s availability for third party partners with royalty free licensing of the SILK codec. Since then it has been incorporated into all subsequent releases of Skype for Windows 4.x, Skype for Mac 2.8 and Skype for Linux. Recently Skype announced that the SILK codec will be supported in the next release of Skype for iPhone.
What does the SILK bring? Basically the best quality “in-your-head” audio available for voice conversations. To recall Jonathan Christensen’s highlights at the time the SILK codec was first introduced, the focus was on:
- improving the audio bandwidth out to 12,000 Hz (12KHz)
- providing bandwidth management to deal in real time with degraded network conditions
- balancing the codec optimization between voice, music and background noise, each of which can have an impact on the overall user experience
- overall robustness to provide a more consistent user experience, regardless of network conditions and an individual caller’s voice signature.
From a user perspective, one will notice crisper conversations, easier interpretation of accents and an overall high quality “in-your-head” voice experience.
But there have been other consequences of SILK’s availability:
- The FREETALK Everyman Headset for Skype ensures that users get the full benefit of SILK by providing an audio chip that effectively replaces the PC’s native audio chip to bring consistently the full benefits of the audio quality available with SILK.
- Similarly the more recently introduced FREETALK Freedom Wireless Stereo Headset also supports SILK’s audio bandwidth.
- A push-to-talk offering, Blabbelon, which provides the ability for gamers to communicate in a multi-party voice algorithm used by gamers, has incorporated the SILK codec, setting a new standard for gaming voice sessions.
But here’s the bigger picture: in order to have a voice conversation that takes full advantage of SILK, the end points of the conversation must incorporate the SILK codec and the telecom infrastructure must support the transmission of SILK conversations.
- The three Skype clients make this possible for end-to-end Skype-to-Skype calls between PC’s (and iPhones with the next Skype for iPhone release);
- Invoking SILK in Blabbelon by default results in every call participant’s taking full advantage of SILK since it is incorporated into the SILK client.
Incorporation into Skype for iPhone will only take advantage of SILK should the other party also be on a Skype client. But:
- SILK is not currently supported in the recently released Skype for Symbian.
- Nor will it be used in the Skype clients being developed for use by Verizon customers due to the need to link the user’s smartphone to Verizon’s Skype gateway using the underlying 3G wireless voice channel.
- SILK cannot be invoked on SkypeOut calls to the PSTN.
- Nor is it supported yet on any other third party smartphone or other end point handset or phoneset hardware.
And therein lies the challenge: how does one propagate SILK such that it becomes available beyond the Skype and Blabbelon worlds? Skype has taken the first step by making the SILK codec available royalty-free and publishing the source code. However, for broadest adoption it needs to become a recognized international standard.
Recently Skype has been working with the IETF’s “CODEC Working Group” as one candidate to be considered in the working group’s effort to identify and standardize a superwideband audio codec that could be widely used across IP-based conversations. In a “Speaking of Standards” post earlier today, Skype releases source code for SILK super wideband audio codec – the details…, Voxeo’s Dan York has provided an excellent detailed background, including both the political and technical considerations leading up to the adoption of such a codec as an IETF standard for interoperability across service providers and end point software and hardware suppliers. He concludes with:
Still to be seen is if there are any additional licensing terms from Skype, but overall this is a great step forward for those of us who want to see better overall audio quality on the Internet. Obviously it’s in Skype’s interest to do this so that the CODEC WG might consider SILK as the codec to standardize on versus the others or creating a new one… but still… it’s great to see. What do you all think?
And it sure helps Skype’s case that Skype has recently hired executives who have IETF working group experience such as Director of Developer Relations, Jason Frischl and Chief Technology Strategist Dr. Julian Rosenberg.
Bottom Line: Ever since the launch of SILK, it’s very noticeable when one is not on a SILK call. SILK can bring significant business productivity advantages when there’s less of “can you repeat that?” or simply misunderstanding of the speaking party. It makes everyone in the family more readily understood during those grandparent Skype video calls. It’s all about improving the end user experience.
There are many efforts to incorporate “wideband (8 Khz)” HD Voice into the telecom infrastructure. GIPS has some excellent wideband codecs; Jeff Pulver is running HD Voice conferences. While HD Voice is definitely an improvement on today’s standard PSTN voice calls it is not up to the quality of SILK-supported calls.
However, spreading SILK’s quality of conversation requires not only the appropriate hardware or client software, as described above, but also support of SILK in the telecom back end infrastructure. Having a standard that every player making the conversation connection possible can work towards would be a major step forward in “clearing up the conversation” when on a voice call.
Update: Skype’s Jonathan Christensen provides an update summary about the SILK codec in his Skype blog post “Advances in audio”. He reiterates the focus he described a year ago:
- 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’s voice signature
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