Speaking of Standards

Want to implement Skype’s SILK super wideband audio codec? You know, the software that makes Skype’s audio calls sound so rich and like you are “right there”?

If so, just check out Appendix A of this massive Internet-Draft:

http://tools.ietf.org/html/draft-vos-silk

Coming in at 566 pages, this I-D is what the word “ginormous” was meant for… but if you look closer, you’ll see that the first 26 pages describe the SILK codec and provide the usual IETF language. The remaining 540 pages are the C source code for a reference implementation of SILK!

This is part of the work of the IETF’s “CODEC Working Group” (see “MOVING INTO THE IETF” below) to standardize on a new wideband audio codec and essentially upgrade the audio quality of the Internet. Let me lay out some background…

LAUNCHING SILK

Last year at eComm 2009, Skype’s Jonathan Christensen announced the SILK codec and that it was being released for free licensing to third-party software and hard developers. Skype made available a developer site focused on the codec:

https://developer.skype.com/silk

Christensen trumpeted these items as the strengths of SILK:

• 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

You can watch his eComm announcement here:

or view his slides here:

As a heavy user of Skype, I can personally attest to the fact that the SILK codec, available in all the latest versions of Skype, really does provide an outstanding audio experience.

MOVING INTO THE IETF

With the SILK codec available, folks at Skype started to look at how the codec could be made an open standard so that more folks would implement it. In May 2009, Skype’s Jason Fischl started a discussion around forming a new IETF working group to produce a standardized wideband audio codec. Much discussion and several meetings followed that did lead to the creation of the CODEC Working Group within the IETF. The group, created in January 2010 and co-chaired by Peter Saint-Andre and Michael Knappe, is open to the public – anyone can join the mailing list – and the problem statement is very clear:

According to reports from developers of Internet audio applications and operators of Internet audio services, there are no standardized, high-quality audio codecs that meet all of the following three conditions:

1. Are optimized for use in interactive Internet applications.

2. Are published by a recognized standards development organization (SDO) and therefore subject to clear change control.

3. Can be widely implemented and easily distributed among application developers, service operators, and end users.

There exist codecs that provide high quality encoding of audio information, but that are not optimized for the actual conditions of the Internet; according to reports, this mismatch between design and deployment has hindered adoption of such codecs in interactive Internet applications.

There exist codecs that can be widely implemented and easily distributed, but that are not standardized through any SDO; according to reports, this lack of standardization and clear change control has hindered adoption of such codecs in interactive Internet applications.

There exist codecs that are standardized, but that cannot be widely implemented and easily distributed; according to reports, the presence of various usage restrictions (e.g., in the form of requirements to pay royalty fees, obtain a license, enter into a business agreement, or meet other special conditions imposed by a patent holder) has hindered adoptions of such codecs in interactive Internet applications.

According to application developers and service operators, an audio codec that meets all three of these would: (1) enable protocol designers to more easily specify a mandatory-to-implement codec in their protocols and thus improve interoperability; (2) enable developers to more easily easily build innovative, interactive applications for the Internet; (3) enable service operators to more easily deploy affordable, high-quality audio services on the Internet; and (4) enable end users of Internet applications and services to enjoy an improved user experience.

Now, here is a critical point:

The CODEC Working Group is *NOT* just about Skype’s SILK codec!

The CODEC WG is working to identify and standardize a wideband audio codec that could be widely used over the Internet. Skype’s SILK codec is one of the potential codec candidates, as shown on this matrix of codecs from the CODEC WG. Another strong contender is the CELT codec from the Xiph.org team. There is a Internet-Draft out now for discussion that specifies the requirements for a standardized codec:

http://tools.ietf.org/html/draft-ietf-codec-requirements

The goal of the CODEC WG is to either identify an existing codec that meets those requirements or create a new codec. Either way, the goal is to have a standardized high quality audio codec that we can all use.

For those interested, the CODEC working group will be holding its first official face-to-face meeting at IETF 77 in Ananheim, California, on March 22nd. A draft agenda has been posted.

SKYPE’S SILK SOURCE CODE

So back on Monday, March 8, a new version of the SILK Internet-Draft was published and announced on the codec mailing list. As the draft notes:

The source code for the reference implementation of the SILK codec is provided in Appendix A. This fixed-point source code is the normative specification of the codec. The corresponding text description in this document is provided for informative purposes. For more information about SILK, the patent licensing terms, and to download the source code in a convenient format, please visit [silk-website]. For any additional questions, please send an email to SILKSupport@skype.net.

This particular phrase is interesting and amusing to me:

This fixed-point source code is the normative specification of the codec.

I translate that as “the source code is the specification”.

If you do read the source code, you see Skype’s license on the code is rather open:

Copyright (c) 2006-2010, Skype Limited. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, (subject to the limitations in the disclaimer below)
are permitted provided that the following conditions are met:
- Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of Skype Limited, nor the names of specific
contributors, may be used to endorse or promote products derived from
this software without specific prior written permission.
NO EXPRESS OR IMPLIED LICENSES TO ANY PARTY'S PATENT RIGHTS ARE GRANTED
BY THIS LICENSE.THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND
CONTRIBUTORS ''AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING,
BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

Essentially it is a Skype version of the permissive “Simplified BSD License“, which I would summarize as a non-lawyer as essentially “keep the copyrights intact, don’t use Skype’s name for endorsement, and Skype is in no way responsible if bad things happen from using this code”.

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?

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The W3C Voice Browser Working Group recently released the fifth working draft of the VoiceXML 3.0 specification at:

http://www.w3.org/TR/2010/WD-voicexml30-20100304/

This newest draft made the following changes:

• Added V2 Convenience Syntax appendix, referenced from convenience syntax section..
• Updated Core concepts (sections 4.1 and 4.2)
• Added section stub for the Subdialog Module
• Added Disconnect Module
• Added Connection Resource
• Expanded the description of the Basic Profile
• Added placeholder for new “Enhanced” profile
• Added Play RC
• Added SCXML code and issue list to Grammar module
• Added Record Module
• Updated language identifier definition and link to BCP47
• Added Property Module
• Added basic Transition Controllers text to the Document Initialization and Execution section

A diff between this draft and the previous draft is available at:

http://www.w3.org/TR/2010/WD-voicexml30-20100304/diff.html

Given that Voxeo’s Dan Burnett is the co-Editor-in-Chief of the VoiceXML 3.0 specification, you can expect to see more postings here about VoiceXML 3.0 as it continues to move along the path toward becoming an actual standard.

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Want to learn about the Session Initiation Protocol (SIP)? Would you like to understand how the SIP protocol works and why it is the dominant open standard for communication today? Want to understand the challenges SIP faces and what’s being done to overcome them?

If so… and if you will be attending VoiceCon in Orlando, FL, March 22-25, you’ll be able to join my (Dan York) 3-hour tutorial on “SIP Fundamentals and Prospects” on Tuesday, March 23rd, from 2-5pm. The abstract VoiceCon has posted is this:

SIP (Session Initiation Protocol) has become the dominant protocol for IP communications. This workshop explains SIP — how it works, the major issues impacting deployments and how SIP will evolve in the future.

The session focuses on the technical aspects of SIP and how it is used. It analyzes in detail the major components of SIP architecture, SIP addressing and registration, session establishment, SIP message routing and connecting SIP across the PSTN. You will learn about SIP extensions and how SIMPLE works for IM/presence. The workshop also examines some of the challenges SIP faces, including NAT traversal (and the tools developed to cope with it: STUN, TURN and ICE) and security. The tutorial concludes with an assessment of how SIP may evolve and its role in peer-to-peer environments. You will receive an inventory of SIP resources—books, papers and organizations.

I’m very much looking forward to the session… although I still do have some work to finish up on the materials. For the past while my friend David Bryan has given these tutorials at VoiceCon events, but given that he also chairs IETF working groups he would need to clone himself since this VoiceCon is the same week as IETF 77 in Anaheim, California. It’s a wee bit hard to flip between coasts… and as anyone who has ever been to an IETF event knows, the meetings are intense and he is needed out there.

If you can’t attend VoiceCon this year, I’ll probably do some SIP tutorial webinars in the future and perhaps you’ll see something popping up over at Voxeo University… stay tuned. And if you are at VoiceCon, please do stop by and say hello… or send me an email in advance letting me know.

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FYI, a new version -08 of my P-Charge-Info Internet-Draft is now available:

http://www.ietf.org/id/draft-york-sipping-p-charge-info-08.txt

For an understanding of what P-Charge-Info is all about, read why I first wrote it, P-Charge-Info and incredible disconnect between PSTN billing and the new world of SIP, and then my update last year on the -07 draft.

Version -08 really only has a minor tweak to the ABNF notation for the “npi-value” and then a new Appendix A clarifying the npi-values and their relation to ANSI T1.113.

I am hoping that I can very shortly request IESG consideration to move this document along the path to being an RFC. The only remaining issue is that my co-author, Tolga Asveren, has brought forward a proposal for simplifying the parameters a bit. I’ve forwarded that proposal to several people I know are very interested in this draft. We’ll see where it goes from there. I’d very much like to move this along soon, so we’ll see.

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As a “security guy“, I have been pleased to watch the emergence of the OAuth Working Group within the IETF and the work that is underway to create an actual IETF specification for OAuth. I haven’t had time to participate, but I’m glad to see that work going on.

If you aren’t aware of OAuth, it’s basically a way that you can authorize a application or service to interact with another application or service on your behalf without giving that first application or service your user ID and password for the second service or app.

For example, if you were a Twitter user in its earlier days, every time you wanted to use another application or web service with your Twitter account, you had to give that app or service your Twitter user ID and password. There’s a security issue here in that you are entrusting your credentials to some other company or application – and trusting that they won’t share those credentials. There’s also a configuration issue in that if you change your password you then have to go to all the other services and provide the updated info. Now, with OAuth support in Twitter, when you want to add a new service to interact with your Twitter account, you are prompted to login to your Twitter account and authorize or deny the access for the new service. The key point is that the new service or application never gets your Twitter credentials. (And as another example, OAuth is what our IMified service uses to allow an automated bot to interact with your Twitter account.)

Anyway, OAuth emerged out of the developer community and now there is work underway in the IETF to create official standard specifications to help in promoting OAuth implementation. As a first step, it was announced this week that OAuth 1.0 will be published as an Informational RFC. As noted in the announcement:

The OAuth protocol was originally created by a small community of web developers from a variety of websites and other Internet services, who wanted to solve the common problem of enabling delegated access to protected resources. The resulting OAuth protocol was stabilized at version 1.0 in October 2007, and revised in June 2009 (revision A) as

published at <http://oauth.net/core/1.0a>.

This specification provides an informational documentation of OAuth Core 1.0 Revision A, addressing several errata reported since that time,

as well as numerous editorial clarifications. While this specification is not an item of the IETF’s OAuth Working Group, which at the time of writing is working on an OAuth version that can be appropriate for publication on the standards track, it has been transferred to the IETF for change control by authors of the original work.

This first step will get a base level spec out so that people looking to implement OAuth will have an IETF specification they can use. The RFC hasn’t been published yet, but the draft that will be an RFC is here:

http://tools.ietf.org/html/draft-hammer-oauth

It’s good to see this work going on within the IETF and I look forward to seeing further work there. From my perspective, OAuth is a great step in helping secure connections betweens apps and services over the web… which is good for all of us as more and more moves into the cloud.

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Over at Ars Technica, author Gilad Shaham has started a series of posts about the SIP protocol. So far the two installments are:

• Wired for sound: how SIP won the VoIP protocol wars
• VoIP in-depth: An introduction to the SIP protocol, Part 1

The first article gives some background about SIP and goes on to explain how SIP prevailed over H.323 as the dominant standard for VoIP traffic today. The second article goes through the details of basic SIP messaging and explains how SIP proxy servers and registrars fit into the picture, complete with some diagrams that nicely explain call flows. The author indicates that the next article in the series will dive into SIP calls in more detail.

If you are new to VoIP or to the SIP protocol, both of these articles are great tutorials that will help you learn more about what SIP is all about. If you are familiar with SIP, you still may find some interesting tidbits mixed into the text. The articles are good to see and I’m looking forward to reading the next installments!

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Registration for the 77th meeting of the Internet Engineering Task Force (IETF) is now open. The IETF 77 event will take place March 21-26 in Anaheim, California (home of Disneyland).

Sadly, it looks like I personally will be missing this IETF meeting as I’ll be across the continent in the other home of Disney, Orlando, for VoiceCon Orlando 2010. I somehow don’t think the cross-country travel will work that week. The good news, though, is that the timezone will be such that I will be able to participate remotely. Remote participation wasn’t really an option for me with the recent IETF 76 in Japan.

One interesting point in the email announcement was that the IETF is offering essentially a “day pass” again for people who either want to just see what IETF is all about or who only care about sessions happening on a particular day. Per the email, for $200 you get:

1. Attend all sessions during any one day of the Meeting, and partake of the food and beverage during the breaks
2. You select which day to attend when you show up onsite to check-in
3. Payments may be made onsite without a late fee
4. Pass can be upgraded to a full Meeting Registration, however, late fee may apply if initial one-day payment not made before Early Bird deadline
5. Attend Sunday Tutorials at no additional charge
6. Attend Sunday Welcome Reception at no additional charge
7. Attend Wednesday and Thursday Plenaries at no additional charge

Seems like a nice option if all you just want to go to a day of sessions.

In any event, more information and the registration form can be found in links from the IETF 77 meeting page.

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So what Twitter accounts are out there that write mostly about industry standards?

Thus far I’ve found:

@rfc
@w3c

I’ve added both to a new Twitter List to which you can subscribe, if you like, at:

http://twitter.com/voxeo/standards

If you know of other accounts, please let me know. (Thanks!) There is an @ietf account out there, but it’s only had one tweet back in November 2008.

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Today the W3C’s Voice Browser Working Group released another draft of the VoiceXML3 specification. This one primarily adds a new section, 6.16, on Speaker Identification and Verification (SIV), part of what is often referred to as “voice biometrics”. If you haven’t been following the development of VoiceXML 3.0, the draft is worth a read, particularly the beginning where it talks about what the W3C is aiming to do with this new version of VoiceXML.

We’ll be writing more about VoiceXML 3 over the months ahead as the draft continues on it’s path toward becoming a specification. At a real high level, the goals the W3C has include:

• Flexibility – more powerful flow control
• Modularity and adaptability – ability to adapt to usage on everything from mobile devices to the IVR servers to network media servers
• Extensibility – ability to be extended as new technologies and communication means become available.

VXML3 will add SIV functionality, support for video and more. As you’ll note in the spec, it adds “profiles” to support different types of features and functions (to get to that “flexibility” goal).

Voxeo’s own Dan Burnett is co-editor-in-chief of the specification and has obviously been heavily involved over the past weeks and months in moving this work along. I’m hoping to get him on video soon giving more of an overview of what’s new in VXML3… stay tuned…

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This week in New York City, Tim O’Reilly gave a keynote at the Web 2.0 event that I definitely put in my “must-see” category. Not because of anything visual… I mean, it’s just Tim standing on stage talking… but because of his message.

There is a war on out there on the Internet.

It’s a war between those who would like to keep the Internet as the open platform for innovation that it has been for decades… those who champion “The Internet Way” – and those who would like to return the Internet to the world of walled gardens from which it emerged. In his excellent piece published on Monday, “The War For The Web“, Tim speaks of the sides as “Small Pieces, Loosely Joined” and, of course, “One Ring To Rule Them All”. He concludes with:

It could be that everyone will figure out how to play nicely with each other, and we’ll see a continuation of the interoperable web model we’ve enjoyed for the past two decades. But I’m betting that things are going to get ugly. We’re heading into a war for control of the web. And in the end, it’s more than that, it’s a war against the web as an interoperable platform. Instead, we’re facing the prospect of Facebook as the platform, Apple as the platform, Google as the platform, Amazon as the platform, where big companies slug it out until one is king of the hill.

And it’s time for developers to take a stand. If you don’t want a repeat of the PC era, place your bets now on open systems. Don’t wait till it’s too late.

This IS the battle that will frame the Internet in the next years. As I wrote a few months ago in ‘Of DDoSs and SPOFs: How Twitter and Facebook violate “The Internet Way”‘, the way of the Internet is to use “distributed and decentralized” services. That’s how email works… that’s how the “web” works… that’s what excites me about the promise of Google Wave – not just that it’s a great platform for collaboration (and as I show here, it is), but that the Wave protocol has been designed from the start to be about federation… to be about distributed and decentralized services.

This war is a large part of why I work here at Voxeo, where one of our core values is “Unlocked Communications“, where we are huge believer in open standards (and chair/co-chair many of the standards committees), where we do things like open source our Tropo cloud telephony platform (“The Cloud Must Be Open!”) and where, in contrast to Nuance and TellMe as Tim mentions at 13:22, we give away our speech recognition engine for free as part of our Prophecy IVR/application platform… that’s why I’m here at Voxeo. It’s a war for openness that I believe we must win!

But listen to Tim… and then ask yourself – which side of the war are you on?

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