Speaking of Standards

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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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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Recently we came across a great series of articles at InformIT on the subject of VoiceXML. Written by Frank Coyne, they cover the range from an introduction to VoiceXML up through using VoiceXML with social networks. Nicely, the author mentions Voxeo and talks about how the exercises he lays out can be done using a free developer account on our evolution.voxeo.com site. Here are the articles:

• “An Introduction to VoiceXML“
• “Part 2: Working with Forms“
• “Part 3: VoiceXML Grammars“
• “Part 4: Grammars, Scope and Event Handlers“
• “Part 5: VoiceXML meets Web 2.0“

You can also get a list of all the articles as well as a blog entry from Frank Coyne back in August titled “Mashin’ Up with Voice XML“. To probably no one’s surprise, I was personally most intrigued by Part 5. Having done a ton of work with XSLT stylesheets in the past I enjoyed the part about creating dynamic Voice XML using XSLT stylesheets to generate the VoiceXML:

http://www.informit.com/articles/article.aspx?p=1017851&seqNum=6

Linking VoiceXML to triggering Gmail delivery is also quite cool:

http://www.informit.com/articles/article.aspx?p=1017851&seqNum=8

As I am personally just coming up to speed on VoiceXML, I’ll be working through many of these tutorials in my own evolution account. (Since accounts are free, you are welcome to sign up and check it out yourself.)

Technorati Tags: voicexml, tutorial, voxeo

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