Archive for the ‘Voxeo’ Category

Want to learn about SIP? Come to my SIP Tutorial at VoiceCon March 22

Thursday, March 4th, 2010

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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New version of P-Charge-Info (08) Internet Draft available

Wednesday, March 3rd, 2010

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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SIPit 25 coming up Sept 14-18, 2009, at UNH IOL

Monday, August 17th, 2009

sipit.jpgSIPit 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… ;-)


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New version of P-Charge-Info (07) Internet-Draft now available – and an update on the draft status

Friday, August 14th, 2009

ietflogo-2.jpgThis 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


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Voxeo’s VoiceObjects acquisition further promotes the open standard of VoiceXML

Tuesday, December 9th, 2008

VoiceObjectslogo-1.jpgAs 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.

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Videos from the IETF 73 standards meeting…

Monday, November 24th, 2008

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:

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

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New revision of MRCPv2 submitted – allows interop with different ASR/TTS engines

Thursday, November 6th, 2008

ietflogo-2.jpgWith 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:

MRCP-simple.jpg

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.

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SIPit 23 coming up Oct 13-17 in Lannion, France – registration closes soon

Tuesday, September 9th, 2008

sipit.jpgThe 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.

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Video: Interview with Dan Burnett on being named 2008 Speech Luminary as “Man of Standards”

Monday, September 8th, 2008

At SpeechTEK in New York City a few weeks ago, our own Dan Burnett was recognized by Speech Tech Magazine as one of the “2008 Speech Luminaries” for all his years of work on industry standards relating to speech. We were delighted for Dan to receive the (well-deserved!) recognition and I had a chance to record a brief video interview with Dan at SpeechTEK:

As Dan mentions, he is Director of Speech Technologies in our Office of the CTO (OCTO) reporting in to our CTO, RJ Auburn, and is responsible for looking at how to constantly improve our speech recognition technology and also ensure it is compliant with standards.

Congratulations, Dan, on the recognition by Speech Technology Magazine!


P.S. And yes, for those following along at home, Dan Burnett and I were both hired into the OCTO at about the same time… we thought about instituting a rule where all new OCTO employees had to be named “Dan”, but thankfully that rule was ignored with the recent excellent addition of Wei Chen!

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P2PSIP and pushing voice down into local clouds…

Friday, May 23rd, 2008

p2psip.jpgWhy do you write so much about P2P SIP? Who’s really going to use it? Why do you care? Isn’t it really just a lame attempt to create an open standards version of Skype?

As you might imagine, I’ve heard those questions more than a few times. And yet still I keep writing about it. Why? Part of the answer lies in my post back in Decembertoday the world of SIP is all really “client/server” but the future might be quite different. Today you have SIP user agents that register and connect to SIP servers (which might be SIP proxies, SIP registrars or other types of servers). In fact, our own Prophecy product is a powerful SIP application server. Our Evolution developer portal is a massive SIP-application-server-in-the-cloud (more here) that connects to and from SIP clients.

But now imagine a SIP deployment without servers.

Imagine that you could simply distribute a series of SIP phones and have them all rapidly interconnect to each other and start communicating. Think of how fast you could potentially deploy a small office. In this time of US presidential campaigns, I think of the “advance teams” that are dropped into some office space in some city to organize an area. Imagine if they could be given a bunch of phones which just self-organized into a working communication environment? Plug the phones into a network switch and have them sort out extensions, PSTN gateways, all of that. There are all sorts of similar situations. Teams of consultants or auditors. Emergency environments. Short-term branch offices. Outside of the rapid deployment, there are just general uses in small businesses that don’t want to pay for servers. You could even see it being used in a home environment. Today this kind of thing can be done with “teleworker” phones hooked back to a central IP-PBX or hosted service (I know because I helped launch one back in 2003), but what if it could be even easier?

This is the promise of P2P SIP. Take a bunch of SIP endpoints and they form their own P2P “cloud” (or “overlay network” in P2P lingo). They discover resources like PSTN gateways or application servers. If a phone dies, the P2P cloud routes around it and continues. Communication happens.

Now the reality is not quite there today. There is a great amount of work being done within the IETF on this subject through the P2PSIP Working Group. There are P2PSIP implementations (mostly open source and a few commercial). There is a great amount of academic research going on (one example here). It is all still early days, though.

And on one level, that is what is so exciting. We are re-imagining what a network could be. We are pushing the power and intelligence truly out to the very edge of the network.

We are creating clouds.

Local clouds. Self-organizing clouds. Network “clouds” that connect to other clouds. It’s a different mindset… a different paradigm… thinking of networks not in terms of servers with their clients but in terms of nodes able to communicate with other nodes.

Will it work on a large scale? We’ll see… there’s a whole whack of security issues… privacy issues… scalability issues… but people are looking at those issues. Skype has certainly shown that a P2P telephony environment can be created and can be quite successful. (Although it should be noted that purists might not consider Skype a pure P2P network because they do have enrollment servers that deal with authentication, etc.) Other P2P networks for file sharing have shown the potential is there to build massively scalable networks. The building blocks are all around us.

So what’s the Voxeo angle, eh? Why do I care about it?

Two answers, really. First, there is the basic reality that just because you are creating a new way of connecting a telephony system it doesn’t remove the need for voice application services. You still want to run voice applications. You still need IVR systems. You still need ways to mashup voice with web services. You still need to connect to the PSTN. Some of those services may be able to live within the actual P2P cloud. Some of them will need to be in external servers or services. Obviously that’s what we do and so my interest is in how our software and services might play in this evolving space.

Second, it’s all about clouds. While there is a huge buzz these days about “cloud computing” and “virtualization“, this is what we’ve been doing here at Voxeo since our founding back in 1999! You can go right now to our Evolution developer portal, create your free account and start building voice applications that run in our massively distributed computing cloud. Behind the scenes, there’s a massively-scalable, geographically-distributed, open-standards-based, redundant (and, incidentally, patented) network architecture that virtualizes your voice and IVR applications. You have no idea what server your voice apps are running on and you don’t care. Need more ports… need more CPU cycles… the cloud adjusts for all of that. It’s all transparent to you.

Our “cloud” is then connected out into other “clouds”. You can connect to our cloud from the PSTN, SIP or Skype and then go back out to the PSTN or SIP clouds. We’re part of the massive interconnect currently being built online.

So as there is the potential to create local P2P “clouds”, my interest on multiple levels is pretty basic – how do we connect those small local clouds to our larger cloud and from there to the rest of the interconnected voice and data networks? I want to look at how our services can enable the proliferation of P2P SIP clouds.

Sure, large numbers of “production” deployments of P2P SIP are probably 3-5 years out… maybe even longer (but maybe not). At the current time there’s not a whole lot I can really do except engage in the ongoing conversations about P2P SIP and try to see what where it’s going. But that’s what I and my colleagues in our Office of the CTO do. We’re the guys up in the crow’s nest looking out and scanning the horizon for what’s coming next. Companies that expect to be around have to keep doing that. So that’s what I do. And that’s why I write about P2P SIP and why I’ll keep writing about it.

It’s all about the clouds… and how we connect them all together…

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