March 09, 2008

Down at IETF 71 in Philadelphia this week

ietflogo.jpgThis week (March 10-14) finds me down in Philadelphia for IETF-71, the 71st meeting of the Internet Engineering Task Force (the people who write the standards for the Internet). I don't honestly know how much I'll be blogging here on this blog. I do hope to be writing some over at the "Speaking of Standards" blog on Voxeo's site. We'll see. These meetings tend to be rather intense.

If you'd like to follow along with what's happening here at IETF, I've written up some instructions about how to join in the audio streaming and IM group chats. I've also posted what I think will be my schedule, which will give you a sense of what the various VoIP-related sessions.

That's the news...

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February 12, 2008

The EComm 2008 Interview with Skype's Jonathan Christensen should be required reading...

42F19C6B-67C5-433E-91B4-641B9323CD48.jpgAs we enter into the final month before eComm 2008, I would suggest that the interview with Jonathan Christensen, Skype's general manager of audio and video, should be required reading for anyone seriously interested in this space. Why? Well, in part because Jonathan Christensen does provide some good information about what Skype has done and is doing but also because it provides some good insight into what one of the people driving Skype's agenda is thinking about this space. Take one of the final paragraphs where he answered Lee Dryburgh's question about what he saw as the the future of communications (bold emphasis added by me):
Well, a big question I guess and, having worked on the space for quite a while, I think that it's only going to get more interesting over the coming years since, well, like this open spectrum for example. You know, I just have to reiterate, I think that anybody who has not figured out that the Internet is the platform and that there isn't any such thing as walled gardens that will survive, or sub-networks [such as AOL tried] that are going to survive, those people are doomed. The intersection of these worlds is going to be chaotic. It's going to be violent. It's going to be messy for a while but it is going to happen, and the Internet will survive as the one open platform. You are going to see a trend towards extreme innovation at the edges - on the devices, in the PC platform, in software, all around the edge of the Internet.

I think that you are only going to see further disruption of the telecom industry and the emergence of totally new businesses that we can't imagine today. I think that [the] net result, that drives me every day, is that we're going to have this very rich, open, cheap and accessible communications. This is going to be not just a game changer for the telecom industry, but will be a change agent for all of humanity. So, a platform that allows us all to see each other and hear each other more clearly maybe makes us a little bit less crazy, less polarized and more open as a world society.

Good stuff... and the whole interview is worth a read. Given my recent criticism of Skype, I'm particularly pleased to read the comments I emphasized in bold. Jonathan Christensen will be giving one of the keynotes at eComm 2008, March 12-14 in Silicon Valley and if you haven't considered going, I would encourage you to do so. It should be a great event!

P.S. I also wrote about this interview in relation to SIP over on Voxeo's "Speaking of Standards" blog.

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February 04, 2008

IETF "RUCUS" BOF to be held about SPIT...

Over on the Voice of VOIPSA blog today I posted about a new session has been approved for the IETF 71 meeting coming up in Philadelphia in March called "Reducing Unwanted Communications using SIP" a.k.a. "RUCUS".Hannes Tschofenig, who submitted the proposal, has created a RUCUS web page and is looking for feedback. I'm planning to be at the RUCUS session at IETF 71 and would encourage others who want to talk about voice spam / SPIT to join in as well!

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January 15, 2008

Skype says "No" to VoIP interoperability - *because customers aren't asking for it!* - Well, I am!

skype_logo.pngSo Skype says that they have no plans for interoperability with other VoIP systems because their customers aren't asking for it??

By way of Dameon Welch-Abernathy today I learned of Phil Wolff's post back in December about ZDNet's interview (Got all that? ;-) with Skype's VP of telecoms, Stefan Oberg. The article was primarily about Skype's London phone number debacle, but this was the part that most irritated me:

Another issue which may concern business users of VoIP is the Enum registry, which aims to unite not only the various VoIP providers — referred to by some as "islands" due to their lack of interconnection with each other — but the entire VoIP and traditional telephony worlds.

Asked whether Skype had considered opening up its famously closed communications protocols, Oberg claimed that there had been no customer demand for interconnection. "[Customers] are not saying they would love to call a VoIP provider on a different network," he said. "Customers are asking for better video and better conference calling. If it is something that customers really ask for, we would consider it, but it is very easy for anyone to get on the island."

Well, Mr. Oberg, here is one paying customer of Skype who can state unequivocally:

"I would love to call a VoIP provider on a different network!"

Here's the thing, Mr. Oberg. There are a whole lot of us out there who are looking to build the next voice communication network. We're looking forward to the day when today's PSTN is just some story that greybeards get together and reminisce about. ("Remember when we used to have to dial numbers? And wait for the connections? And remember how much we had to pay our phone companies for the privilege? And remember those 'busy signals'?") We're looking to make it simpler and easier and so that ultimately voice just smoothly fits in to our communication as one of the several different ways we communicate. (others being text/IM, video, etc.)

The funny thing is that many, if not most, of us experimenting with what will be next are Skype users. Probably in many cases paying Skype users since we have Skype Credit and SkypeIn numbers. Because, like you said, Skype makes it "very easy for anyone to get on the island." You do a lot of things right. You've got a very simple and easy-to-use client. Your directory is good. Your use of wideband audio usually gives outstanding audio quality. Your ability to work from very different network environments and through firewalls is great. Some of us love that everything you do is encrypted. You work across the major computing platforms. You make a great product and because you have hit a critical mass with so many of us there, we like to use your product.

But... with statements like this you're living in the same delusion that Facebook has been in until recently. You see, there's this wee tiny little problem:

You are NOT the only island!

Sure, you're probably the largest island with the most parties and easiest docks to land at. But there are a lot of other islands out there. Some of them are other services with whom you admittedly compete. Some are startups. Some of them are the traditional carriers now offering VoIP services to consumers and businesses. A lot of other islands are the companies and organizations now wiring themselves up with IP-PBXs or using back office software from Microsoft or IBM to "voice-enable" their infrastructure. Ditto for some cities and towns that are doing the same thing. In some cases, those islands are wiring voice so far into their business processes and systems that it's truly amazing.

Now some of us, seeing all these islands out there, say... "Hmmm... why don't we just connect the dots?" Let's build some bridges or high-speed ferries between those islands. Let's get them talking together. Let's interconnect the islands and build the new infrastructure. Let's bypass the old PSTN and build the new voice network entirely across the Internet. Let's forget all about those geographic boundaries... let's let voice flow to wherever wherever someone can get an IP address. Anywhere. Anytime. Let's interconnect business systems with other voice systems.

And you know what? We're doing it. Slowly. Very slowly at times. But we're doing it. We're using protocols like SIP and RTP and all the many others coming out of the IETF. We're creating "mashups" and using XML flavors like VoiceXML and CCXML to weave voice into the web. We're starting the interconnection. We're enabling businesses to connect to each other and dial each other directly. We're using SIP trunking to let local systems make and receive phone calls from other parts of the world. We're giving people their choice of endpoint... they can use a range of "hard" phones (traditional pieces of hardware) or "soft" phones (like you are). People can ring my deskphone simply by calling "dyork@corpsip.voxeo.com" using their SIP phone... goodbye hard-to-remember telephone numbers... hello user names.

Oh, sure, we've got lots of problems still to work out. Security is a huge one. You are extending your trust boundary out to include other networks. How do you know they won't send you tons of voice spam? Or abuse your network? Or run up bills on your dime? Privacy is another one. How do you show others only the information you want to? Sooner or later the various governments and tax authorities are going to wake up and realize how badly they are going to be screwed out of revenue by all we are doing - and we're going to have to deal with that. We've still got to agree on how to do certain features between systems. We've got a lot of work to do.

But we're doing it. We're rewiring the phone system. We're creating a new one, not shackled by its history.

The question for you all at Skype seems to be whether or not you want to help build that larger interconnected world. Or whether you want to just hang out on your island and hope that if you throw big enough parties and advertise "Free Beer" enough that everyone will forget about their own islands and just come over and join yours.

You know what? You'll get a lot of people to come on over. Today. And probably for some time. You've got a fun island to hang out on.

Meanwhile, the rest of us will keep on with our rewiring and remixing. We're building the fabric of what comes next. We're coding the DNA for the future of voice. We'd love it if you joined us. I'd love it. It would be great if I could call my colleagues on SIP extensions from directly within Skype. Not through some Skype-to-PBX gateway that really winds up running multiple instances of Skype... but through an actual SIP gateway. I'd love it if I could give them a SIP address like "dyork@sip.skype.com" that they could use to call me on my Skype client wherever I was. You know, I'd probably wind up using my Skype client more if I had that capability! You have a great UI. Why shouldn't I add my SIP contacts there, too?

What SIP contacts you say? Yes, clearly I'm an "early adopter"... one of those geeks who goes around chasing bright shiny objects. Guilty as charged. But each day what I do is becoming easier and easier for others to do. And you know what? If you supported SIP contacts, those of us who talk and write about topics like this would probably do a lot to evangelize you. We'd actually help you with your marketing.

Now you do make this excellent point:

"In order to provide richness, we have to create our own protocols," Oberg added. "SIP and the standard [VoIP] protocols simply can't do it."

You're right. Almost all the traditional vendors in the VoIP space do use their own proprietary protocols to give the rich communication experience people want. Cisco. Nortel. Avaya. Alcatel. Mitel. Others. But you know what? Their hold on the market is being disrupted. Lots of new players coming in. Big ones like Microsoft and IBM - who are interestingly supporting the open standards we're using. So the traditional vendors are evolving, too. They're supporting SIP for interconnection. Sure, they still have their parties on their islands and show people how great it is there, but they do allow bridges to be built. They understand the need to interconnect.

You're right, too, in that SIP only supports basic calls. We know that. We're working on it. So come join us. Join the IETF mailing lists. Send someone to IETF 71 in Philadelphia in March. Advocate for how we should interconnect to you. Building the Interconnect is long, often glacially-slow work, full of many people with different agendas, many of whom will all disagree. Join with us. You'll lose some battles and win others. But together we might just have a chance at making it all happen.

Or... just keep hanging out on your island throwing parties and trying to attract new people. Maybe it will work.

In the meantime, please don't say that customers aren't asking for interoperability. Count me as one who is:

"I would love to call a VoIP provider on a different network!"

I bet if I ask around, a few of the people I know would like that, too.

If you read this far, thank you for listening. You can now return to your island. Meanwhile, we've got some rewiring to do...

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November 29, 2007

Introducing "Speaking of Standards", a new Voxeo blog about industry standards, IETF, W3C, SIP Forum, etc.

200711292028A large part of why I have NOT been writing here all that much in the past few weeks is that I've been busy in my new role with Voxeo working on a corporate blog portal. I've been covering a bit of that odyssey over on my Disruptive Conversations blog as well as in my weekly reports into the For Immediate Release podcast. It's been a great amount of work but also a lot of fun - I've been very lucky to have a colleague who does amazing things with CSS and graphics, and so the sites look a whole lot better than they would if I were left to my own devices.

I'm very pleased to say, now, that we've reached the point where I'm willing to link to our work and talk a bit about what we are doing. The main blog portal is the predictable "blogs.voxeo.com" but the weblog that we're really starting to use and could be of interest to readers of this blog is our "Speaking of Standards" blog found at:

http://blogs.voxeo.com/speakingofstandards/

I've obviously been very occasionally writing here about standards and some of that may continue, but I expect most of my writing on the subject will now occur over on this new Voxeo weblog - and I'll naturally be writing more on the subject. We'll be writing about the IETF and SIP standards, but also the W3C and standards such as VoiceXML and CCXML that I've never covered at all here. We'll be linking to events and tutorials we find and generally providing whatever information we can about standards affecting our industry, as well as Voxeo's views and implementations of those standards.

Why would Voxeo sponsor a weblog about standards? Primarily because the company and our products are all about open standards - which was one of the things that attracted me to the company after they first approached me. I've since learned that they've been leading the IVR industry in adopting open standards. As the products page says in the "Fast Facts" sidebar:

  • 100% Standards based IVR
  • Supports W3C VoiceXML 2.0
  • Supports W3C CCXML 1.0
  • Supports W3C SRGS 1.0
  • Supports W3C SSML 1.0
  • Supports CallXML 3.0
  • First platform with XML call control
  • First platform with XML conferencing
  • First shipping CCXML implementation
  • First SIP/VOIP IVR platform

Not bad, eh? Add to that the fact that our CTO (my manager), RJ Auburn, chairs the W3C's Working Group on CCXML and we've hired other folks involved with standards efforts... all of that is why we added a weblog on standards.

So if you would like to see our view on industry standards, find tutorials about various standards or learn about standards-related events we may be attending, I would invite you to come on over and check out "Speaking with Standards" - or subscribe to the RSS feed. While I (and others) will still be working on improving the site, it's mostly done and I'm delighted to be able to return to writing more. Let us know what you think!

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November 19, 2007

I'll be out in Vancouver Dec 2-7 for the 70th meeting of the IETF.

200711191406Just confirmed travel plans today - I will be heading out to the 70th meeting of the Internet Engineering Task Force (IETF) in Vancouver, British Columbia, Canada, from December 2-7. If any readers will be out there (either for the IETF or in Vancouver in general), please do drop a note and let me know. This will be my first meeting in my new role with Voxeo and I'm very much looking forward to renewing old acquaintances and also getting more directly involved with the work of the IETF.

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November 12, 2007

Did you know RFC 4733 had replaced/obsoleted RFC 2833 for DTMF signaling in SIP?

Did you know that RFC 4733 replaced/obsoleted RFC 2833? I just learned this myself through a SIP Forum mailing list exchange the other day. For those not aware, RFC 2833 and now 4733 define methods of carrying DTMF signals (and other similar signaling) in RTP streams separate from the main audio component of the RTP stream. A typical example of use might be where you were using a highly-compressed audio codec for audio between two SIP endpoints where the high degree of compression might make it challenging for the DTMF tones to be correctly interpreted on the receiving end. Using "RFC 2833 compliant" signaling, the sending SIP endpoint would send those DTMF tones as separate packets within the RTP stream.

My key takeaway from learning about RFC 4733 is that we should really be talking about "RFC 4733 compliant" signaling... but given that the industry is really only now starting to really talk about "RFC 2822 compliant" signaling, I'm not sure I expect to see that happening anytime soon.

Anyway, here's the abstract from RFC 4733 - you can naturally read the rest of the document to understand more:

This memo describes how to carry dual-tone multifrequency (DTMF) signalling, other tone signals, and telephony events in RTP packets. It obsoletes RFC 2833.

This memo captures and expands upon the basic framework defined in RFC 2833, but retains only the most basic event codes. It sets up an IANA registry to which other event code assignments may be added. Companion documents add event codes to this registry relating to modem, fax, text telephony, and channel-associated signalling events. The remainder of the event codes defined in RFC 2833 are conditionally reserved in case other documents revive their use.

This document provides a number of clarifications to the original document. However, it specifically differs from RFC 2833 by removing the requirement that all compliant implementations support the DTMF events. Instead, compliant implementations taking part in out-of-band negotiations of media stream content indicate what events they support. This memo adds three new procedures to the RFC 2833 framework: subdivision of long events into segments, reporting of multiple events in a single packet, and the concept and reporting of state events.

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April 19, 2007

IETF approves RFC standard for adding dialstrings to SIP

In the usual (and ongoing) flurry of IETF announcements, there was one notice that caught my attention.  It announces that an Internet Draft document about "dialstrings" has been approved to become a standards-track RFC.  So what, you say?  Well here's a bit more info:

This document provides a way of incorporating a dial string into the SIP or SIPS URI scheme. A dial string is a cousin of a telephone number, but rather than taking the form of a fully-qualified E.164 or national-specific telephone number, it is a description of a literal set of dialed digits that would be delivered over a POTS line. As such, it may include pauses, omit prefixes like area codes, and its applicability is necessarily restricted to a particular context (an enterprise, a LATA, etc). Support for dialstrings was formerly a feature of the tel: URI scheme specification (back in RFC2806); since that functionality did not make it into the revision (RFC3966), it is provided here specifically for the SIP and SIPS case.

Think of it as extra digits you have to type when making a call... or extra keys you have to press to start a service.  The challenge is that SIP proxies and other services need to know that it is a string of numbers that should be handled in a special manner, rather than just thought of as part of a SIP address or something like that.  I mention it here only because it's one of those really low-level things that you can do on the PSTN but until now haven't had a (standard) way to do in SIP. It's also one that ultimately anyone implementing SIP will need to implement.  No RFC number yet, but that will come soon.  Note the nice security warning:

Dialstrings exposed to the Internet may reveal information about internal network details or service invocations that could allow attackers to use the PSTN or the Internet to attack such internal systems. Dialstrings normally SHOULD NOT be sent beyond the domain of the UAC. If they are sent across the Internet, they SHOULD be protected against eavesdropping with TLS per the procedures in [RFC3261].

Yep... as we've been saying over at VOIPSA and Blue Box, you definitely need to think about encrypting SIP if you are sending it across the Internet.  If not, bad things will happen eventually.

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April 05, 2007

Attaining BLISS... (at least in the world of SIP)... a.k.a. why can't we all just get along?

 So you'd like your SIP phones to all work together, eh?  And you'd like your SIP phone from Vendor A to work with the SIP phone of Vendor B and yet give you the business functionality that you used to have in the PBX from Vendor C?

Good luck.

Yes, they will (or should!) all work together for basic call functions, but if you want to do more than just the very basics, you rapidly wind up in the realm of incompatible SIP implementations.  Different vendors support different RFCs... or interpret RFCs differently.  It's a challenge to go beyond basic functionality.

Enter "BLISS", one of the latest working groups coming out of  the IETF. It stands for "Basic Level of Interoperability for SIP Services" and, as noted in its charter, the intent is to define a basic set of functionality ("minimum interoperability requirements") to allow SIP endpoints to interoperate on 4 specific telephony services:

  1. Bridged/Shared Line Appearance (BLA/SLA)
  2. Call Park/Pickup
  3. Do Not Disturb (DND)
  4. Call Completion to Busy Signal/Call Completion on No Reply

More details are on the charter page.  These are just the initial four issues chosen to be addressed and Internet-Draft documents are already circulating on some of the items.  I see it as a necessary group if we are to actually have real interoperability between SIP endpoints, and I commend the group organizers on scoping it to an initial 4 issues (rather than making it wide open).

If you work with SIP, I'd encourage you to check out the BLISS web site, read the "problem statement" and charter... and then join the mailing list (and please read the archives to see what's happened so far).  I've joined the list... and would encourage others to consider doing so.  If we do want real SIP interoperability, we need to hammer some of these things out.

For more information, you can download the set of slides presented by Jonathan Rosenberg of Cisco at the IETF 68 meeting stating the problem and why the BLISS working group is needed.  To make it easy to see them, I'm going to embed a SlideShare version here:

See you on the list...

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April 02, 2007

So who will be first vendor to implement VoIP over RFC4824?

So with the release yesterday of RFC4824, The Transmission of IP Datagrams over the Semaphore Flag Signaling System (SFSS), one has to wonder... which of the vendors will be the first to attempt to implement VoIP transmission in this medium?     I think it would make for a rather slower conversation, but it would certainly be intriguing.  Hmmm... I wonder which would be faster - this method?  Or the avian method defined in RFC2549.  Probably this one, methinks. 

Oh, you have to love a standards body with a sense of humor...

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  • Dan York, CISSP, is Director of Emerging Communication Technology at Voxeo Corporation. He is also the Best Practices Chair of the VOIP Security Alliance (VOIPSA).

    Note that neither Voxeo nor VOIPSA have any connection to this weblog and any opinions stated here are entirely Dan's.

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