Speaking of Standards

While the slides and recording of today’s webinar on IPv6 will be available tomorrow on the Jam Session web page, I also made this set of slides available on SlideShare. The nice thing about SlideShare is that I can also embed them here in this post… enjoy!

Would you like to learn more about IPv6 and how it relates to creating communications applications? Want to understand more about how IPv6 does or does not impact the SIP protocol?

If so, you may be interested in a free 1-hour session I’m giving on Thursday, May 5, 2011. The topic is:

IPv6 and How It Impacts Communication Applications

It’s part of Voxeo’s “Developer Jam Session” series and is targeted at a technical audience. I’ll briefly cover IPv6 basics, talk about how it impacts communication apps, the SIP protocol and then have some demonstrations of SIP-over-IPv6. You can learn more about the session and register on the Jam Session web page.

If you attend live on Thursday you’ll be able to ask questions during the session. If you can’t attend, the session will be recorded for later viewing. (And if you register, we’ll make sure you know as soon as we post the recording.)

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.

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.

SIPit 25 starts four weeks from today at the University of New Hampshire’s InterOperability Lab (IOL). What is SIPit? As the UNH-IOL page for the event says:

SIPit’s, or Session Initiation Protocol Interoperability Tests, are weeklong events where people bring their SIP implementations to ensure they work together. SIPIT Events are open to anyone with a working SIP implementation. The goal of the events is to refine both the protocol and its implementations. The SIPIT events are a driving force shaping SIP into a globally interoperable protocol for real time Internet communication services.

Basically, they are a place where vendors can privately test their SIP implementations against each other. Results of the testing are not publicly released – other than an aggregate news release talking about what occurred overall. It’s a place where, as a vendor, you get a great chance to see how well your SIP-based product interoperates with that of other vendors. It’s also a place where vendors will often bring early implementations of new SIP standards to test those against other vendors working on early implementations. All in all, it definitely helps with moving us all along the path toward increasing SIP interconnection.

We’ll have a Voxeo team at this SIPit. We’ve been based on SIP since we started our company back in 1999 and we’re continually looking at ways to increase our performance and support for evolving SIP standards. We value the feedback we gain from these SIPit events and try to attend when we can.

You can attend, too, as there is still space available. The UNH IOL event page has more info and there is an online registration form as well. (Deadline to register, though, is September 4th.)

P.S. And yes, since yours truly lives about two hours west of UNH, I *am* planning to head over and meet our testing team for dinner probably in beautiful Portsmouth, NH… I’m in “marketing” now, so they don’t let me near the test equipment. I mean, in my world, all the tests just work, right? And they have really pretty charts to go with them…

This morning I submitted a new version -07 of P-Charge-Info. Not much changed in the draft. Primarily I updated the IPR language to use the language of the IETF Trust as of February 2009 and added a reference pointing to RFC 3968.

The P-Charge-Info draft has been rather delayed on its path to becoming an Informational RFC primarily because just as I was about to request “expert review” per the process in section 4.1 of RFC 3427, the SIP and SIPPING working groups decided to revisit RFC 3427 and restructure how changes are made to SIP in general.

The result has been “RFC 3427bis”, a.k.a. draft-peterson-rai-rfc3427bis and section 4 defines a new lighterweight process for registering SIP headers. (Well, it speaks of a “Designated Expert” process, but my criticism of the existing draft is that it doesn’t easily explain what someone has to do to register a new SIP header.) While this RFC3427bis draft has not been ratified as an RFC, the RAI area is proceeding as if it has been and has already replaced the SIP and SIPPING working groups with SIPCORE and DISPATCH.

At this point I’m done with P-Charge-Info in that I’ve incorporated all comments that people have had and all I am looking to do now is move it through the end of the process to an Informational RFC. I will be contacting the RAI Area Directors shortly to sort out exactly what the next step is to get this moving along. In my ideal world, I’d like to see this published by the end of this calendar year.

I do, though, still welcome comments, so if you have any please feel free to pass them along.

P.S. For info about why I originally wrote this draft, see “P-Charge-Info and incredible disconnect between PSTN billing and the new world of SIP

As a strong supporter of open standards, I think one of the aspects of our acquisition of VoiceObjects that was only touched on briefly in the video podcast I did is what this means for the world of standards. Specifically VoiceXML. What intrigues me about VoiceObjects’ platform is not just it’s support for standard VoiceXML, but even more its support for cross-platform VoiceXML. Quoting from our news release:

VoiceObjects uniquely enables the development of phone applications that can be deployed on a wide variety of VoiceXML platforms. This capability is in stark contrast to vendor-specific development solutions offered by Voxeo’s competitors. These single-vendor solutions restrict application deployment to a vendor’s own VoiceXML platform, denying the freedom of vendor independence and application portability the VoiceXML standard was designed to support. Voxeo will continue to openly and actively support VoiceObjects’ application deployment on multiple VoiceXML platforms including Aspect, Avaya, Genesys, Intervoice and Nortel. VoiceObjects will also be available in extremely cost-effective on-demand and on-premise offerings bundled with Voxeo’s own Prophecy VoiceXML Platform.

While VoiceXML is itself an open standard, the way VoiceXML is implemented on some platforms (or “extended“) can certainly wind up restricting deployment to that one platform. We firmly believe customers should be able to develop completely standard VoiceXML apps that can avoid proprietary vendor lock-in and can be moved to other platforms if necessary. I’m delighted to see VoiceObjects added to our portfolio of tools so that even more developers can create voice applications and can be independent and portable. Obviously, we think we have an extremely compelling case for people to deploy their voice applications on our platform, but we want people to do so because it is their choice to do so, not because they are forced to do so.

I’m looking forward to writing more here in the weeks and months ahead about VoiceObjects and the standards it supports.

Technorati Tags: voxeo, voicexml, standards, voiceobjects

I don’t know how many of you reading this “Speaking of Standards” blog are also reading our blog portal, subscribing to the main Voxeo blog feed or following us on Twitter, but if you aren’t, I thought I would mention here that I recently launched a new video blog, “Emerging Tech Talk” and that last week three of my shows were specifically about the 73rd meeting of the Internet Engineering Task Force (IETF) held last week in Minneapolis, Minnesota. The shows were:

• Emerging Tech Talk #004, introducing the IETF

• Emerging Tech Talk #005, about the RAI area, SIP, P2PSIP Security and more
• Emerging Tech Talk #006, with a wrap-up of the week and mention of NAT for IPv6 and a bit about ENUM.

I also recorded some video interviews there at IETF 73 with Eric Burger of the SIP Forum, David Bryan of P2PSIP fame and Peter St. Andre of the XMPP Foundation. I’m aiming to put some of those up soon.

The Emerging Tech Talk video blog will not necessarily focus on standards… it will really be about whatever bright shiny objects I happen to be chasing at the time… but sometimes standards will factor in to it. You are definitely welcome to subscribe to the show feed if you would like to stay up on what I am doing – and comments are definitely welcome. (And yes, I will be making it available as a “podcast” that can be subscribed and downloaded to your computer, iPod or iPhone.) If you are a YouTube user, you can also subscribe to our “voxeovideos” channel directly inside of YouTube.com

Technorati Tags: voxeo, danyork, emergingtechtalk, ett, sip, ietf, ietf73

With IETF 73 coming up shortly in Minneapolis, those of us here in Voxeo were very busy last week getting our Internet-Drafts updated in time for Monday’s submission deadline. One of the major pieces of work was done by Dan Burnett with his new revision of the Media Resource Control Protocol Version 2 (MRCPv2) draft.

MRCP is actually a fascinating protocol to me (okay, admittedly, I’m a standards geek) in that it provides an open standard that allows a system to very easily interoperate with different “media processing resources” such as Automatic Speech Recognition (ASR) or Text-To-Speech (TTS) engines. This is how, for instance, our Prophecy product is able to easily use different ASR or TTS engines. In a very simplified view, it looks something like this:

where the “MRCP Client” is, in our case, Prophecy. Now the cool part about this is that if you need a specific ASR engine for a task, if you can find an “MRCP-compliant” engine it should be able to easily interoperate with Prophecy. Say, for instance, that you needed speech rec for a language we didn’t support, a special TTS engine or something like that.

Anyway, the new draft of MRCPv2 is out there and goes into this in an extraordinary amount of detail. If you do have any comments, by the way, Dan Burnett is open to hearing them (his email address is at the end of the draft).

P.S. If you’ve used our Realtime Debugger or our Prophecy Log Search feature inside of Evolution, you’ve no doubt seen a bunch of messages related to MRCP – this is all part of the communication between our main execution environment within Prophecy and the various ASR and TTS resources being used to execute your application.

Technorati Tags: MRCP, IETF, Standards, ASR, TTS, SIP

The next SIPit SIP interoperability testing event, SIPit 23 will be October 13-17 over in Lannion, France. More information about the event – and the registration info – can be found on the ETSI page for SIPit23 (ETSI is the host for this SIPit.) We’re a big fan of interoperability test events like this because in the end they only help all of our products grow stronger and help SIP advance as a communication protocol. We’re currently planning to have someone from Voxeo over there… perhaps maybe even more than one.

If you have a SIP-based product, do consider attending – and maybe we’ll be able to test some interop together when we’re over there.

Technorati Tags: sip, sipit, sipforum, voice, voip, voxeo