W3C releases 7th Working Draft of VoiceXML 3.0 specification

August 31st, 2010 by Dan York

w3clogo.pngAs the VoiceXML 3.0 specification makes its way toward release, the W3C Voice Browser Working Group announced today the seventh working draft of the VoiceXML 3.0 specification at:

http://www.w3.org/TR/2010/WD-voicexml30-20100831/

This newest draft listed only these changes:

  • Revised Legacy profile description to match current thinking.
  • Removed SIV Resource (section 5.4) since it is now covered along with the recognition resource in section 5.3.
  • Update section 4.4 (Event Model) to match our current thinking about DOM events as the underlying model for all flow control
  • Cleaned up text in sections 1, 2, and 5

Per Voxeo’s Dan Burnett, co-editor-in-chief of the specification, the biggest change is the third bullet about the updates to the event model (section 4.4).

Personally, I’ve found the best way to understand the changes is to view the “diff” of the version from the previous version at:

http://www.w3.org/TR/2010/WD-voicexml30-20100831/diff.html

Given our interest in VoiceXML and VoiceXML 3, we’ll be continuing to write about it here… you can see related stories at:

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


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SIPit 27 announced for Nov 15-19, 2010, in Taiwan

August 31st, 2010 by Dan York

SIPit27.jpgThe SIP Forum today announced that the next SIPit interoperability test event will be held November 15-19, 2010, in Taiwan, Taipei.  A website for the event is now up at:

http://www.etsi.org/plugtests/SIPit27/SIPit27.htm

As I’ve written about in the past and recorded a video interview about, these SIPit events are critical, in my opinion, to helping drive the overall adoption of SIP and open standards in communication systems.

If you are a creator of software or hardware devices that use the SIP protocol, SIPit events are a great way to test how well your equipment works with other SIP implementations. There is a fee, but it’s small for the week-long testing you get to do. More info can be found on the registration page.

P.S. You’ll note that this SIPit event is the week following IETF 79 in Beijing, China… so if you can make the travel work, it’s a great way to combine two weeks of open standards / SIP – related events.


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Want an IETF-branded T-shirt? hat? coffee mug?

August 17th, 2010 by Dan York

ietft-shirt.jpg

Have you wanted a T-shirt, hat or mug to help promote the IETF and your support of the organization?  Now you can buy one, courtesy of a new online store the Internet Society has opened up:

http://www.cafepress.com/ietf

I think it’s very cool to see because many people who attend IETF meetings have wanted to help promote the great work of the IETF… but haven’t been able to have an easy way to do so like wearing a shirt (except for the shirts they receive at IETF events).

Not being a personal fan of white T-shirts (I have young kids), I naturally wanted to see a black shirt… and while they are not there yet, I’m sure they will be at some point.  It’s from Cafe Press and all created “on-demand”, so the number of items is really only limited by ISOC’s choices as well as their ability to upload appropriate graphics (for example, black shirts do need a special image).

I’m sure they’ll be getting all sorts of feedback in the next few weeks with requests for products.  Kudos to ISOC for setting this up!

 


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Registration opens for IETF 79, Nov 7-12, in Beijing, China

August 13th, 2010 by Dan York

ietf-shadow.jpg

News came out yesterday that registration is now open for the 79th meeting of the Internet Engineering Task Force (IETF) to be held November 7-12, 2010, in Beijing, China.  You can find out more and register at:

http://www.ietf.org/meeting/79/

The announcement contains more information and pricing (as does the web site).

Tsinghua University, the host of the meeting, also put together a dedicated website providing more information and including a video promoting Beijing:

http://www.ietf79.cn/

This is a milestone for the IETF in that it is the first IETF meeting to be held in China.


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W3C “Cheatsheet” – great way to look up HTML, CSS

July 27th, 2010 by Dan York

W3Ccheatsheet-1.jpgHave you ever been coding in HTML or CSS and found yourself wondering exactly what the attributes were to a given element? Or wondering what values an attribute can have?

If so, the W3C offers this great “cheatsheet” at:

http://www.w3.org/2009/cheatsheet/

where you can simply enter a search string and look up info about HTML, CSS, SVG or XPath.

For instance, say I wanted to know more about the <div> tag in HTML, I can just enter it and then get a result with links to learn more about the possible attributes or to click over and read the actual specification:

W3Ccheatsheet-div.jpg

The site nicely provides suggestions as you type. For instance, I started typing “align” and received these suggestions:

W3Ccheatsheet-typeahead.jpg

As you can probably guess from the tabs in the screenshots, the site also provides information about development for mobile browsers and guidance around accessiblity, internationalization and typography. A great resource for anyone who is coding up their site. Visit the cheatsheet at the W3C’s site to try it out yourself.


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Two new SIP-related IETF Working Groups: LSD and SALUD

July 15th, 2010 by Dan York

ietf-shadow.jpgThis past week brought the announcements of two new SIP-related Working Groups out of the IETF. In the long-standing tradition of having entertaining names, they are naturally “LSD” and “SALUD”.

The first announcement was for the “Loosely-coupled SIP Devices” Working Group (LSD) covering a topic of personal interest to me – disaggregated media. From the charter description:

Disaggregated media refers to the ability for a user to create a multimedia session combining different media streams coming from different devices under his or her control so that they are treated by the far end of the session as a single media session.

Generally, a given participant uses a single device to establish (or participate in) a given multimedia session. Consequently, the SIP signaling to manage the multimedia session and the actual media streams are typically co-located in the same device. In scenarios involving disaggregated media, a user wants to establish a single multimedia session combining different media streams coming from different devices under his or her control. This creates a need to coordinate the exchange of the those media streams within the multimedia session.

There are a number of existing mechanisms that can be used to coordinate different devices under user’s control and to involve them in the call (e.g. Message Bus (Mbus) [RFC3259], Megaco [ITU-T H.248.1] and SIP 3pcc [RFC3725]). However, these mechanisms are intended to be used in “tightly coupled” scenarios. The use of all those mechanisms requires the presence of a “master” device. That is, at least one among the different devices under the control of the same user must support the control mechanism and be able to become a controller for the other devices in the call. Moreover, the “master” device is supposed to remain involved in the user’s session for its entire duration given that performing a handover of the master role is typically cumbersome and sometimes impossible.

The objective of this working group is to develop a framework for disaggregated media in “loosely-coupled” scenarios, where no single device needs to remain in the session for its entire duration and no single device needs to act as a master. Coordination among devices in this type of scenario is less tight than in the scenarios described above since they do not assume central elements with complete knowledge of the whole media session. While the framework may describe how to use existing mechanisms (e.g., the SIP REFER method) to coordinate devices, the working group will not develop new device coordination mechanisms. The framework may identify the need for new (non-device-coordination) mechanisms to enable the implementation of loosely-coupled scenarios. In case the need for such new mechanisms is identified, the working group will specify them.

For those interested in participating or monitoring the discussion, there is a LSD mailing list to which anyone can subscribe.

The second announcement was for the “Sip ALerting for User Devices” (SALUD) Working Group. This group is looking to tackle an area of interoperability around the semantics behind one part of SIP signaling. The description provides these examples:

RFC 3261 allows a user agent server to provide a specific ringing tone, which can be played to the calling user, as a reference (e.g., an HTTP URI) in the Alert-Info header. In some situations, the calling user may not be able to understand the meaning of the ringing tone being played. For example, a user from a given country may not be familiar with the tone used to indicate call waiting in another country. Similarly, a hearing impaired person may prefer to get a visual call waiting indication instead of a call waiting tone.

RFC 3261 also allows a user agent client to provide a reference to a specific alerting tone, which can be played to the called user (e.g., tones for emergency announcements sent over PBX systems or over the local telephone network). As in the previous examples, in some situations, the calling user may not be able to understand the meaning of the alerting tone being played.

This Working Group is going to be working to define a standardized set of URNs that can be used in the Alert-Info SIP header to pass along the meaning rather than a specific URI of a media file to be played (for instance). This would allow different systems to provide the most appropriate alert to the user based on local or personal settings.

Again, there is a mailing list available for anyone interested in participating in or monitoring the developments of the WG.


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Slides: Standards Update on VoiceXML3

July 1st, 2010 by Dan York

At the Voxeo Customer Summit 2010 last week, Dan Burnett, who heads up our communication with standards bodies, gave an update on what is going on with VoiceXML 3. The slides are now available on SlideShare:

I may have a video recording of this session available later this month if the video turned out okay. (It’s in a long post-production queue.)

Interested in trying out VoiceXML today? Read our VoiceXML tutorials and then either sign up for a free account in the cloud on our Evolution developer portal or download a free version of our Prophecy platform


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IETF 78 registration open – July 25-30 in The Netherlands

June 11th, 2010 by Dan York

ietflogo-2.jpgThe 78th meeting of the IETF is coming up July 25-30 in Maastricht, Netherlands. Registration is open. As was the case for IETF 77, there is also a day pass option available for people who just want to attend for the day.

It does not look like my schedule will allow me (Dan York) to participate directly in IETF 78… it’s the week right before SpeechTEK New York and for us at Voxeo SpeechTEK is really the largest show we participate in… and so I expect I and my team will be quite busy that week. Schedule permitting, I’m hoping to at least catch a few of the sessions remotely.


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If the BBC covers XMPP and the real-time web, it must be mainstream…

May 14th, 2010 by Dan York

xmpplogo.jpgWe were pleased to see the venerable BBC covering the Extensible Messaging and Presence Protocol (XMPP) in an article:

How to make the web work in real-time

The well-done piece provides a nice overview of XMPP and its role in the evolving “real-time” web. Here’s the intro:

That ask-and-you-will-receive mechanism breaks down when too many people want data from a server at the same time. Everyone knows the frustration of websites that will not respond because they are overwhelmed.

A change is under way to help the web cope with this growth and help some parts of it cope better in a real-time age.

Despite the complicated name, XMPP does a very simple thing. Whereas the web, via HTTP, asks for information on behalf of an individual, XMPP makes sure everyone is told.

If HTTP is a performance for one, XMPP is a stadium gig.

The article goes on to talk about use cases, companies using XMPP, how XMPP impacts search and much more.

It’s great to see this coverage of XMPP by the BBC and the article is quite right. We are in a time of transition.

Since the early days of the web in the early 1990’s, the dominant communication model has been one of “polling”. You had to check a website to know if there was a change. You had to refresh/reload your browser. The emergence of RSS feeds certainly helped with this… now a RSS reader (or your web browser) could “subscribe” to a RSS feed … but still, your RSS reader had to check whether there were new items in a feed.

While this approach has worked, it doesn’t scale terribly well as anyone who has tried to visit a congested website can attest. We’re in the midst of a transition from the “polling” model to a “publish/subscribe” model. In the new model, content providers publish information and consumers subscribe to that information… but in an important change:

the consumers are notified when there is new content.

That is the critical difference.

The XMPP protocol plays an important role in this change, and understanding XMPP and how it works is critical for anyone looking to understand the technologies underlying the “real-time web”. Here at Voxeo, we’re big fans of XMPP and it is a major component of our IMified service as well as our ClackPoint and Tropo services. Great to see the coverage by the BBC and we look forward to seeing where XMPP and other realtime protocols will take us all in the time ahead…


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Explaining why RTP does not mandate a single security mechanism

May 14th, 2010 by Dan York

ietf-shadow.jpgGiven my background in security, I’ve been asked a number of times… why didn’t the IETF or someone just mandate ONE way to secure audio or video sent via RTP? After all, RTP by itself does not have any real security built-in and while Secure RTP (SRTP) is increasingly used, it’s not universal.

I’m obviously not alone in being asked this question, and Colin Perkins and Magnus Westerlund at the IETF wrote up an Internet-Draft that was recently approved to be an Informational RFC on just this topic:

Why RTP Does Not Mandate a Single Security Mechanism
http://tools.ietf.org/html/draft-ietf-avt-srtp-not-mandatory

From the introduction:

The Real-time Transport Protocol (RTP) [RFC3550] is widely used for voice over IP, Internet television, video conferencing, and various other real-time and streaming media applications. Despite this, the base RTP specification provides very limited options for media security, and defines no standard key exchange mechanism. Rather, a number of extensions are defined to provide confidentiality and authentication of RTP media streams and RTCP control messages, and to exchange security keys. This memo outlines why it is appropriate that multiple extension mechanisms are defined, rather than mandating a single security and keying mechanism.

The document goes on to list where RTP is use and issues with various security mechanisms (including SRTP). If you ever wondered why everyone doesn’t just use SRTP… this short document may be well worth the read.


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