Speaking of Standards

While at SIPNOC 2011 last week in Herndon, VA, I shot a good number of photos of the presenters and attendees. I’ve uploaded some of the better shots to our Flickr account and you can view the SIPNOC set there:

I tried to capture most all of the presenters there. The images are licensed under a Creative Commons Attribution license which essentially means that you are welcome to use them as long as you link back to the photo (if online) or provide an attribution link in written material.

If you are in any of the photos and want a higher resolution version for your own usage, please just email me and can get one to you.

For photography geeks, almost all of the photos were taken with my Nikon D90. A few were taken with my iPhone 4. Most were not altered but a few had the “Enhance” edit done in iPhoto.

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.

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

How many conferences have you attended where the WiFi network – if there even is one – has been really poor? Or charged you an arm and a leg to use it?

That doesn’t happen at IETF meetings… bandwidth is usually decent and accessible in in all meeting rooms and common areas – at no charge to meeting attendees. Why? Because the IETF brings in its own network!

Indeed, there’s even a document on “Meeting Network Requirements” which spells out how to arrange such a network. Here’s the abstract:

The IETF Meeting Network has become integral to the success of any IETF meeting. Building such a network, which provides service to thousands of heavy users, spread throughout the event venue, with very little time for setup and testing is a dramatic challenge. This document provides a set of requirements, derived from hard won experience, as an aid to anyone involved in designing and deploying future networks.

If only other conferences could have a network like this!

P.S. Here’s a piece in the Philadelphia Business Journal that goes into this: “1,500-strong laptop invasion to hit Marriott for Net task force” (hat tip to Comcast’s IETF71 blog).

Technorati Tags: ietf, conferences, wifi

On Sunday night I head down to Philadelphia for the IETF-71 meeting for the whole week. It will be a crazy week full of discussions and conversations about all the various standards under development. The RUCUS BOF I’ve mentioned before will be on Monday as is the SIPPING Working Group. MEDIACTRL Working Group (of key interest to us here at Voxeo) is on Wednesday as is SPEERMINT and PEPPERMINT (Hey, it’s IETF, you have to have cute names!). Thursday brings SIP, BEHAVE, AVT and ENUM and Friday morning winds it all up with the P2PSIP working group.

Being who I am, I’ll pretty much sit in all of the “Realtime Applications and Infrastructure” (RAI) working groups as sometimes activity in one group turns out to have great relevance to work in other groups (or to work here at Voxeo). I’ll be online the Jabber chat rooms probably much of the whole time as well.

If you’ve never seen the full agenda for an IETF meeting, it’s pretty incredible (at least to me!). In any given timeslot there are typically eight simultaneous meetings of various working groups, BOFs, research groups, etc. This makes sense if you remember that the IETF is developing standards for pretty much all aspects of the Internet. While I usually never leave the world of RAI, there are groups dealing with security, DNS, email, IPv6, network routing, time (seriously!), host configuration and pretty much every other subject you can imagine relating to the Internet. Take a look!

And yes, the days do begin with a breakfast at 8am and meetings that go until 7pm (often with additional ad hoc meetings afterwards). The good news is that the breaks between sessions usually have food and drink to keep you recharged.

For those attending who wish to stalkfind me, here below is the agenda I think I’ll be following (subject to the fact that it can, of course, change). Like I said earlier, it’s pretty much all of the RAI area.

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MONDAY, March 10, 2008
0800-0900 Continental Breakfast – Franklin Hall Foyer

0900-1130 Morning Session I

RAI

mmusic

Multiparty Multimedia Session Control WG

1300-1500 Afternoon Session I

RAI

rucus

Ruducing Unwanted Communications using SIP BOF

1520-1720 Afternoon Session II

RAI

ecrit

Emergency Context Resolution with Internet Technologies WG

1740-1950 Afternoon Session III

RAI

sipping

Session Initiation Proposal Investigation WG

TUESDAY, March 11, 2008
0800-0900 Continental Breakfast – Franklin Hall Foyer

0900-1130 Morning Session I
One of these:

IRTF	rrg	Routing Research Group
OPS	v6ops	IPv6 Operations WG
RAI	geopriv	Geographic Location/Privacy WG

1300-1500 Afternoon Session I

RAI

bliss

Basic Level of Interoperability for SIP Services WG

1520-1720 Afternoon Session II

RAI

avt

Audio/Video Transport WG

1740-1840 Afternoon Session III
One of these:

IRTF	asrg	Anti-Spam Research Group
RAI	simple	SIP for Instant Messaging and Presence Leveraging Extensions WG

1850-1950 Afternoon Session IV

RAI

xcon

Centralized Conferencing WG

WEDNESDAY, March 12, 2008
0800-0900 Continental Breakfast – Franklin Hall Foyer

0900-1130 Morning Session I

RAI

mediactrl

Media Server Control WG

1300-1500 Afternoon Session I

RAI

speermint

Session PEERing for Multimedia INTerconnect WG

1510-1610 Afternoon Session II

RAI

peppermint

Provisioning Extensions in Peering Registries for Multimedia INTerconnection BOF

1610-1700 PGP Session
(Yes, I’m one of those people who does actually go to PGP key signings.)

pgp

PGP Key Signing

1700-1930 IETF Operations and Administration Plenary – Salon G/H

THURSDAY, March 13, 2008
0800-1700 IETF Registration – Franklin Hall Foyer

0800-0900 Continental Breakfast – Franklin Hall Foyer

0900-1130 Morning Session I

RAI

sip

Session Initiation Protocol WG

1300-1500 Afternoon Session I
One of these:

IRTF	hiprg	Host Identity Protocol
SEC	saag	Security Area Open Meeting
TSV	behave	Behavior Engineering for Hindrance Avoidance WG

1510-1610 Afternoon Session II
One of these:

RAI	avt	Audio/Video Transport WG
RAI	enum	Telephone Number Mapping WG

1700-1930 Technical Plenary – Salon G/H

FRIDAY, March 14, 2008

0800-0900 Continental Breakfast – Franklin Hall Foyer

0900-1130 Morning Session I

RAI

p2psip

Peer-to-Peer Session Initiation Protocol WG

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Over on the EComm2008 blog, Lee Dryburgh posted the transcript of a fascinating interview with Jonathan Christensen, general manager of audio and video at Skype. The interview is well-worth a read as Jonathan provides a preview of his upcoming keynote at EComm 2008 with his view of Internet-based communication and talks about advances they have made at Skype with regard to wideband audio and echo cancellation. I do definitely agree with his statement around the improvements they’ve made with echo cancellation on the Mac. Ever since upgrading to the latest Skype, I’ve made many calls with it from my MacBook Pro without any headset whatsoever and have been told the quality has been excellent (and it has been for me when I’m talking to other headset-free Skype users).

Much more relevant to this blog, though, were Jonathan’s statements regarding SIP. At the beginning Jonathan mentions how he originally got very excited by the vision of SIP and ran around stirring up interest at Microsoft where he worked then. But at the end of the interview, Lee asked Jonathan to elaborate on his earlier comments about SIP. This is what Jonathan said:

Yes, so just one clarification – we use SIP. Where, by comparison to the other operators, we are one of the largest SIP users in the world. All of our SkypeOut minutes and SkypeIn minutes traverse the PSTN via SIP interfaces, basically. So, we use it as an interop protocol where we need to.

I think that the vision of the early SIP founders has been largely unrealunrealized [See comments] in the SIP world. SIP is typically just used for these very mundane trunking applications, like the one that we have, or sending calls between two networks and it’s just calls. The vision of multi-modal communications and rich end points has largely failed within the same. I think that a big part of this is that they didn’t pragmatically just solve basic problems like NAT traversal, for example. They also evolved the specification to the point that it no longer had its lightweight appeal. So, we’ll see, SIP will continue to be [the] dominant protocol in terms of this sort of narrowly defined scenarios but I think that, when it comes to rich communications, you are going to see more of this fragmentation. You’re going to see some islands of providers who are just solving the problems. Just making it work for the user and not being religious about the protocol for example.

Has the vision of rich communication over SIP been “largely unrealized”? What do you think? Are his statements true? Or exaggerated?

FYI, if you are attending EComm 2008 you’ll have a chance to hear Jonathan Christensen’s keynote directly. And if you aren’t yet attending EComm 2008, why not?

P.S. For the record, we, too, are huge users of SIP for our connections to/from the PSTN and also throughout our hosted Evolution platform as well as our on-premise Prophecy product. Developers on our hosted platform also get by default SIP *and* Skype dial-in numbers for their applications.

Technorati Tags: skype, sip, ecomm, ecomm2008, conferences, jonathan christensen