The IETF usually streams much of their conferences so people who are not free to travel may still participate. Quite often there’s an audio stream, sometimes there’s video and a web share of any slides. They usually stream from several of the meeting rooms, as multiple sessions are typical going on in parallel.
This coming Monday’s Technical Plenary is also going to be the basis of an experiment. The session is going to be streamed via WebRTC. That means that anyone with a WebRTC capable browser will be able to monitor the session. It further implies that the session on Opus will in fact be streamed using Opus…which seems only fitting.
The session will be recorded for those who cannot participate live. Since 5:40pm is Berlin is 10:40am in Houston I’m hopeful that I may be able to list in using Chrome.
While I have been basically offline for the past week, I took some time while awaiting one of my flights home to read some news. That little exercise revealed that the Freeswitch community call this past week featured Phil Zimmermann describing VoIP encryption and more specifically his ZRTP protocol. Happily, the recording of the call was put online Thursday.
Phil is of course one of the leading lights in the world of encryption. The call features Phil speaking plainly and openly about the need for encryption and the manner of its implementation in ZRTP.
The call remains a community call, so it goes off in various directions at times, including a little Asterisk bashing. However, Phil makes a good effort to keep the call informative, making it a great listen for anyone interested in voice security.
On the mailing list of the IETF’s CODEC working group Jean-Marc Valin made a significant announcement on Feb 4th. It reads as follows:
We’d like to announce that the Opus codec is now ready for testing. The bit-stream is now is a “pseudo-freeze”, which means that unless a problem is found during testing/review, there are no longer any changes planned. The only exception to this are the SILK-mode FEC and the stereo SILK mode, which should be landing in the next few days. Considering that these are not critical features, we felt like the testing phase could already begin.
Please recall that OPUS is the new codec arising from the combination of CELT and Skype’s SILK. It’s multiple operating modes accommodate many different applications, from extremely low-latency high-quality links between production studios, to voice applications on very low bit-rate channels. OPUS brings us the current state-of-the-art in audio codec technology in a royalty-free, open source form.
The samples and explanations provided are first rate. They clearly illustrate the merit in ultra-low-latency for some applications, as well as exemplify how CELT currently fares against other common codecs at a variety of bit-rates and with various types of source material.
As someone who’s passionately involved spreading the gospel of HDVoice I’ve been following the mailing list of the IETF CODEC Working Group. They’ve been working towards a new IETF RFC on a brand new wideband codec standard.
Starting with four submissions from Broadvoice, SpiritDSP, Skype and Xiph.org I think that they have made startlingly good progress over the past year. The group has actually arrived at a solution that provides for a codec that is a hybrid of SILK and CELT. It was recently announced on the mailing list that this new hybrid codec is to be known as “Opus.”
At IETF78 in Nagasaki, Japan the working group met to further their efforts. There’s a good recording of the session that, amongst other things, gives considerable detail about the hybrid nature of Opus.
There has been close co-operation between developers at Skype and Jean-Marc Valin of Xiph.org, such that they already have sample code running and have conducted some structured listening tests. The results of the listening tests are reported to be excellent.
It’s very interesting how the hybrid codec works. It can leverage both CELT and SILK principles acting on different frequency bands to generate the most optimal audio for a given bit-rate. It supports a wide range of sample rates from 8 KHz (PSTN narrowband) to 48 KHz (production quality audio) and bit-rates from 8 kbps to 128+ kbps per audio channel.
Opus may have a huge role to play in our transition away from the legacy PSTN. It’s very encouraging to hear that the CODEC Working Group is progressing so quickly.
Much is being made of the recent events in the IETF CODEC Working Group . Specifically, the fact that Skype has included the c source code for their SILK codec in the Draft RFC document.
Dan York has some excellent coverage including a good general backgrounder on the matter. Jim Courtney has also posted something interesting, as has Phil Wolff of Skype Journal.
A lot of what is being expressed seems to me unbridled enthusiasm for what is seen as a bold, even surprising move on the part of Skype. I agree that this is a gutsy move…and one that I applaud. However, I’m also here to reign in the enthusiasm just a bit. Tempering it with a dose of reality we can see this in a larger context and keep our eyes on the larger goal…ubiquitous wideband telephony.