Posts categorized "VoIP"

Skype takes a SIP of Cisco with UC500 Skype For SIP certification

skypeforsip.jpgIt's been a busy month for the folks in the Skype For SIP project. First, back on September 9, Skype announced ShoreTel interoperability. Then last week on September 17, Skype announced interop with the open source SIPFoundry sipXecs product.

Today, though, is Skype's biggest announcement yet - they are announcing the certification of Cisco's Unified Communications 500 Series for Small Business as interoperable with Skype For SIP.

Beyond simply the interop, what's perhaps more interesting is to note the direct Cisco involvement with this news release (through a quote). Looking at the overall industry, it's interesting to see Cisco and Skype connecting. I admit that I haven't studied Cisco's UC500 product much at all, although per the news release it sounds like they are doing some interesting things with it:

The Cisco Unified Communications 500 Series platform is part of Cisco’s Smart Business Communications System which continues to expand having just added a new set of IP phones with high definition audio, a unified threat management device as well as support for third party application integration, including products from healthcare, automotive and insurance industries.

Congrats to both Skype and Cisco on this announcement. I expect we'll be seeing more of these announcements in the weeks and months ahead as Skype continues to aggressively court partners. The Skype For SIP offering does provide some useful functionality for on-premise IP-PBX systems:

  • Ability to receive inbound calls from Skype users
  • Ability to receive inbound calls from PSTN users through "online numbers" (formerly SkypeIn)
  • Ability to place outbound calls to PSTN users

The Online Number functionality is particularly interesting as you can easily set up any series of numbers in other parts of the world that ring back into your IP-PBX. Sure, you can do that with any other SIP trunking provider, too, but Skype makes it incredibly easy to provision those numbers - and for a very low annual cost, too. Making your IP-PBX accessible to all the Skype users, too, is quite powerful.

Now if only you could make outbound calls to Skype users... (NOT possible with Skype For SIP, but possible with Skype For Asterisk).

Anyway, congrats again to Skype and Cisco on this announcement.


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I'll be at VON in Miami on Monday, Sept 21, 2009

von2009logo.jpg If any of you will be down at VON next week in South Beach, Miami, FL, next week, I'll be part of two presentations on Monday, September 21, 2009. The full abstracts are outlined on a Voxeo events page, but the titles are:
10 – 11:15am, Beyond Boxes: The Future of the PBX 11:30 – 12:30pm, The Apps Race: Building a Developer Community in the New Telecom World

The second one should be fun as it's with my good friend Thomas Howe (who also has a spiffy new website). It's just Thomas and I and a moderator, talking about developer ecosystems. Good stuff!

I'm only there at VON on Monday. That evening I'll be driving up to Orlando where I'll be spending Tuesday through Friday at Voxeo's corporate office. But if you are down at VON, please do say hello.

P.S. And yes, this is the "new" VON put on by Virgo Publishing after they purchased the VON name and tradeshow from Pulvermedia. It will be interesting to see how it is as a show.


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I'll be out at ITEXPO this week in L.A.

itexpo-logo-1-1.jpgIf any of you are heading out to ITEXPO tomorrow through Thursday in Los Angeles, I'll be there on Wednesday. As I note on a Voxeo events page, I'll be speaking twice, pretty much back to back:

9:30 – 10:15am, Exploring Applications in the Cloud

11:00 – 11:45am, SIP Trunking and Security

The first is a panel discussion that should be quite interesting. The second is another version of the VOIP / SIP Security talk that I've been giving at Ingate's SIP Trunking Seminars for the past few years (and that always seems to be popular). More details and session abstracts on the events page I set up.

I'm looking forward to catching up with many friends at the show, including Andy Abramson, who I haven't seen for a while.

If you will be out there, please do say hello.


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Silicon Alley Insider: How Comcast Ate Vonage's Lunch

Fascinating chart out of Silicon Alley Insider today showing the incredible growth Comcast has had in terms of IP phone subscribers versus that of Vonage:

I doubt the chart is a huge surprise for anyone following the industry, but it still does make for an interesting graphic. Despite all the advertising money that Vonage can throw out there, Comcast and the other cable providers have the inherent advantage that they can easily offer powerful "triple-play bundles" of cable TV, Internet access and phone service.

Comcast is no longer my provider (Time-Warner services Keene, NH, where I live.), but when I lived in Burlington, VT, we had Comcast for Internet access and Verizon for phone (and we didn't have anyone for TV, since we don't watch it). The offers that Comcast kept sending us, though, encouraging us to switch, were quite compelling. The amount I paid for Internet access would have been lower if I had either phone or TV with Comcast, and even lower if I had all three.

If we actually watched TV and therefore wanted cable TV, the economics of the "triple-play" would be very hard to beat... so it's no surprise at all to me to see this chart. I would expect we'll continue to see the growth of one and the continued stagnation of the other.

What would be more interesting to me would be to see the subscriber growth of Comcast versus the other MVNOs and the DSL providers. Fascinating times we live in...


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For a brief bit - Skype video calls from 30,000+ feet on Southwest Air WiFi!

Upon entering the Southwest Airlines plane this morning on my flight to Orlando to visit Voxeo's headquarters, I immediately noticed a "WiFi zone" sign on a column by the entry door. Naturally, I had to pull out my Macbook Pro and give it a whirl... and, given the issue of Skype-blocking last summer... try out Skype.

To my great surprise and pleasure, it worked great. (For a little while - see below.) Here's Bruce Lowekamp:

03_33 | Call with Bruce Lowekamp-1.jpg

As you may or may not be able to see in the graphic, our call had been up for 3 minutes and 33 seconds when I snapped the picture.

Technical Call Info-1.jpgUnfortunately, I didn't have easy access to my headset, so I couldn't really have a great conversation. The ambient background noise in the plane was really too much for my voice to be heard unless I bent down toward the microphone... and likewise even with the MacBook Pro volume up all the way it was a bit hard to hear Bruce unless I bent down toward the laptop.

And, of course, I didn't really want to annoy my fellow passengers. :-)

You can see on the right the technical stats for the call. Packet loss was surprisingly low. At the time I captured this window, I had a 3% receiving packet loss... but through much of our call that was hovering down around 0%. A roundtrip of 789ms is, of course, rather entertaining.

Sadly, though, the connectivity didn't last forever. After about maybe 30 minutes of trying this out and making video calls to different people, my Skype account went offline.

COMPLETELY offline.

No instant messaging/chat, either. Which is exceedingly annoying.

Out in Twitter-land, Fred Posner asked Southwest if they were blocking VoIP, to which Southwest replied:

@fredposner yep...we do.

Bummer.

Actually, what annoys me more is Southwest blocking Skype IM. Other than the novelty today, I don't know that I really want to be making calls from the plane... although I could see the usefulness from time to time.

What I do want is the Skype IM... since I use Skype IM heavily for communication with people including my team. It seems, though, that my Skype connection is completely blocked. Which is unfortunate, given that all my other IM services seem to be working fine.

Southwest, can you please figure out how to block the Skype voice and video, but still leave us the chat?

Despite the VoIP-blocking, though, it was seriously great to be able to use my laptop and work online during the flight. Even better, on this flight the Southwest WiFi was free while they are apparently testing it out. I did a couple of speed tests from DSL Reports, naturally, and seemed to be getting around 3Mbps down and around 200-250 Kbps up. Here's one of the results:

Speed Test - dslreports.com.jpg

What a crazy world we're now in... 3 Mbps downlink... on a plane!

P.S. And of course I had to snap a picture like this:

danonsouthwest.jpg

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Digium takes on the "fax issue" with Fax For Asterisk...

digiumlogo.gifI can't stand fax. I can't. It's a technology that I just wish would go away. It kills me that fax is one of the main reasons I didn't drop my landline in my move. Yet the reality is that fax usage is everywhere... and probably will be for quite some time if for no other reason than the complete and utter simplicity of fax usage. Print out your message, or write your message (you know... that thing we all used to do... take a writing tool (pen, pencil, crayon, charcoal, etc.), grasp it in your hand and make marks on some writing surface...), just stick that message in your fax machine, punch in the number and press Send. It's hard to get much simpler than that.

But the lack of fax has been a barrier to many a premise-based IP-PBX deployment. Everything's going great... people are looking at all the great things they can do with VoIP and Unified Communications, etc. They are figuring out distributed architectures that are all IP-based. It's all looking really cool technically and will save money, too. All is going well and then someone asks "What about the fax machines?" And so people wind up with kludge solutions using analog breakouts or local lines or attempting Fax-over-IP or keeping some TDM around or... or... or...

We as an "IP communications industry" have to figure out a way to address the "fax issue" if we really do want to build our big interconnect and move beyond the PSTN into a richer communication experience. (And the SIP Forum, by the way, has formed a FoIP Task Group to look at this issue.)

In this context, I was intrigued to see that this week the folks at Digium announced a new service called (Duh!) "Fax For Asterisk". Here's the standard blurb:

Digium's Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry's leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium's Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.

Translation: You can send and receive faxes through Asterisk using either TDM or Fax-over-IP (T.38), licensed on a per-channel basis.

In a rather smart move on Digium's part, they've also rolled out "Free Fax For Asterisk" where you can get a free 1-channel license for Fax For Asterisk for an Asterisk installation. This will let people at least play with FFA and may be all that some small offices need.

I'm naturally intrigued by the FoIP side of the offering, which the FAQ dives into in a bit more detail. Unfortunately I've been doing some work on my home network and my Asterisk installation isn't operational right now, but I expect I'll be bringing that back online soon and expect I'll then be experimenting with this a bit more as well. So if I could get termination with a SIP service provider who offered T.38.... could I at least solve the receiving side of the picture? (And yes, I can also solve that through eFax and a zillion other providers.) The sending side still requires some hardware changes on my end to scan it in...

Now if we can just make it ultimately as simple as sticking the papers to send in the document feeder and pressing Send...


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Skype on the iPhone: Some initial thoughts...

Unless you remained under a rock last week, you know that Skype was released for the iPhone, somewhat predictably rocketed to the #1 downloaded iPhone app, shot past 1 million downloads in two days and then just today went past two million downloads. As Skype's Peter Parkes' writes:
I’m back with an update – and it’s no surprise that we’ve stormed past the 2 million download milestone. This means an average rate of 2-3 downloads per second since the app first appeared on the 31st March.

2-3 downloads per second! You have to think Skype's got to be rather happy about those numbers. Jim Courtney had a good post rounding up some of the Skype for iPhone coverage.

The good news, for me, was that Skype for the iPhone did include support for persistent group chats, which I noted was my one big desire for the iPhone client.

I've been using the client now off and on for the past week, and thought I'd write down a few initial impressions:

  • Audio quality - I've not actually used it for too many calls, but when I have the audio quality (over WiFi) has been great. I'm looking forward to trying it out more whenever I next travel. Interestingly it seems to use the G.729 codec versus any of the ones that Skype developed themselves.

  • Group chats are supported, but are buggy - As I said above, the good news is that group chats, both public chats and also non-public group chats, are supported by the iPhone client. The bad news is that the groupchat support is still a bit buggy (and yes, I realize that it's a 1.0 release). I found a wacky behavior (which I reported) where the iPhone client would only show you the 10 last updates in a group chat, which made it kind of useless.

    I've also noticed that when I go into some group chats in the iPhone client, they don't have the most recent updates in them, which you could expect... but they don't ever seem to update. Now, this could be part of the known issue Skype has with multiple Skype clients simultaneously using the same Skype ID. The updating of groupchats between multiple Skype clients sharing the same ID is a bit funky with regular PC/Mac clients, so it's probably no surprise that it could be strange with the iPhone client in the mix. I'll have to try it sometimes without being signed into Skype anywhere else.

    Still, the great thing is that I can update group chats while I am mobile and, if things work right, also catch up on what's been written in those chats.

  • Battery life - Somewhat predictably, I've found that keeping the Skype For iPhone app running does eat up the battery a good bit. At one point when I left it sitting on my desk for a while purposely, I did see that the battery went down considerably - and my iPhone got a bit warm, too. Now perhaps this is because I do have so many chats open. I did, however, expect this to be the case. I can see myself using the Skype For iPhone client primarily for dipping into the Skype flow to send messages or catch up on what's going on. I don't see myself using it for an extended period of time while mobile... although of course I can always plug it in.

All in all I've been quite pleased with how the Skype For iPhone client works in my initial usage. I need to use it while traveling to really get a sense of how well it performs. It does, of course, suffer from the two inherent issues in the Apple iPhone platform: 1) you can only use the voice portion over WiFi; and 2) there is no background processing - any app has to run in the foreground. Of these, #2 is really the larger annoyance because you can't be doing something else on the iPhone and still receive Skype IM messages or calls. Still, it's great to have a Skype client on the iPhone.

What do you think? Have you installed it and tried it out? What have you found works well? Or doesn't work well?


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Skype tears down more walls with "Skype For SIP"

NOTE: I have a few updates to the post that I am putting at the bottom of the text.


skype_logo.pngWould you like Skype users to be able to call your business' phone system? Would you like to connect your phone system to Skype's network and make use of their cheap calling rates? If you have an IP-PBX or other call server that supports the Session Initiation Protocol (SIP), you may now have those options.

For a company that only a bit over a year ago was saying that customers weren't asking for interconnection, today Skype has done something rather dramatic and lowered their walls a bit more with the announcement of the beta program of "Skype For SIP". With this announcement from the "Skype For Business" group, companies with SIP-enabled phone systems will be able to receive calls from Skype users - and make calls using Skype's network at Skype rates. The news release (and blog post announcement) highlights these four aspects:

  • Receive and manage inbound calls from Skype users worldwide on SIP-enabled PBX systems; connecting the company Web site to the PBX system via click-to-call
  • Place calls with Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX; reducing costs with Skype’s low-cost global rates
  • Purchase Skype’s online numbers, to receive calls to the corporate PBX from landlines or mobile phones
  • Manage Skype calls using their existing hardware and system applications such as call routing, conferencing, phone menus and voicemail; no additional downloads or training are required

Let's take those one at a time - and then take a look at some details and what's missing.

[NOTE: For the ease of writing this post I am going to refer to the SIP system as an "IP-PBX", but it could be a "call server", "call manager", "application server" or anything else than can send and receive SIP signaling. It could be open source or commercial - that doesn't really matter.]


INBOUND CALLS FROM SKYPE USERS

In a pre-announcement briefing, Chris Moore, a senior product manager at Skype, indicated that when the "Skype For SIP" service is fully released, the current "Business Control Panel" will be revamped a bit and will have an area where you can sign up for the service, identify your IP-PBX and associate one or more Skype names with your IP-PBX.

Calls from Skype users to those Skype names would then be routed across the SIP connection to your IP-PBX, where the IP-PBX would deal with those incoming connections exactly as it would any other incoming SIP connection. Call routing will be handled in the IP-PBX. Perhaps the incoming call will go to an auto-attendant, IVR, call-center software or other application. Perhaps it will be routed to a person. From the IP-PBX point-of-view, it's just another incoming call.

skypeforsip1.png

From a Skype users point-of-view, they are simply calling another Skype ID. It's a free call that seems just like any other Skype call.

You will, Moore stated, be able to associated multiple Skype IDs with your single SIP connection. Now I'm not sure what kind of call info you get across the SIP connection, but if you do get to see the Skype name the person is calling you could do some interesting call routing based on the Skype ID called. For instance, if someone made a Skype call to "companyname-help" it could be routed one way and calls to the Skype ID "companyname-sales" could go another way.

In any event, this aspect of the service makes it so that any Skype user can call your IP-PBX.

SIP TRUNKING, SKYPE-STYLE

The second aspect of the "Skype For SIP" service is that you can use Skype's global network for connections out to the PSTN... what we have been generically calling "SIP trunking" for some time now. As part of the Business Control Panel registration you will apparently indicate which Skype ID is to be charged for outbound calling (what we used to call "SkypeOut") and then any outbound calls via SIP will be charged to that account.

Effectively, Skype just opened up cheap international calling to businesses everywhere using Skype's cheap rates.

skypeforsip2.png

The already crowded "SIP trunking" market just got another big player. Configure your IP-PBX to send calls out across Skype's network to the PSTN... and start calling. Right now the plan as I understand it is that you would pay the regular SkypeOut rates, without the subscription plans that are available to individuals. Skype's Moore did say that they may evaluate some form of plans during the beta period. As we see all these various details, we'll have to see what impact this will have on the existing players.

ESTABLISHING PHONE NUMBERS AROUND THE WORLD

The third aspect of the announcement is that you can establish phone numbers around the world that will route to your IP-PBX via the Skype For SIP connection. Since, as I mentioned above, you are associating one or more Skype accounts with the SIP connection to your IP-PBX, you can also associate the "Online Numbers" (what we used to call "SkypeIn" numbers) of those Skype accounts with your IP-PBX.

So if you want a phone number in one of the 20 countries where Skype currently supports Online Numbers, you just buy the number for one of the Skype accounts connected to your IP-PBX.

For $60 per year per online number.

Not a bad deal if you want phone numbers in other area codes or other countries that will ring your business' phone system. Again depending upon what kind of caller ID and called ID info you receive, some interesting call routing might be possible.

skypeforsip3.png

(This assumes, of course, that Skype keeps the current pricing for Online Numbers and doesn't charge any additional costs.)

USE EXISTING HARDWARE

Skype's point here is really just that there's no additional software... it's just an inbound SIP connection to your IP-PBX that you deal with in the same way that you deal with all other inbound SIP connections.


COMMENTS/CONCERNS

Based on the conversations I had with folks from Skype about this new service, I do have a couple of comments and concerns:

SECURITY

Yes, okay, you would expect this of me. Obviously we'll have to wait to see the implementation, but it sounds like Skype has thought this through a good bit. They'll support the standard SIP digest authentication, but more interestingly they will support restricting connections based on IP addresses. If your IP-PBX - or more likely the SIP-aware firewall or SBC or edge proxy - has a fixed address you will apparently be able to enter this in and use that to limit inbound SIP connections. Skype also indicated that when they service moves out of beta into full production they intend to support TLS-encrypted SIP as well.

Similarly, on the media side the beta will support regular RTP but Skype is looking to support SRTP in the full production release.

CODECS

In a somewhat bizarre move (to me), Skype is initially releasing this in the beta program with only support for the G.729 codec. For those who don't follow audio codecs (used to encode the audio of your voice to send across the network), the G.729 codec compresses audio and results in lower bandwidth usage. It also, unfortunately, requires the payment of royalties which typically then require the purchase of additional licenses from the IP-PBX vendor. (This is true even in the case of Asterisk, where you can purchase G.729 licenses from Digium.)

Given that the very people likely to want to use Skype's services for low-cost calling are also the same people who are probably not going to pay for G.729 licenses, it seems that there is a bit of a disconnect here.

The good news is that the folks at Skype say that they are going through the testing to make the much more widely used (and royalty-free) G.711 codec available and expect to have that ready within the first couple of weeks of the beta program.

On the wideband side, Skype folks indicated that at the point in time where there are SIP endpoints that support the SILK codec, which Skype recently said they will make available in binary form for free, those SIP endpoints should be able to make and receive calls to/from Skype users with wideband audio.

(I'll just note that Skype's rationale when I asked them about why G.729 vs G.711 was that they currently use G.729 with all their many SIP termination providers and so using that codec just seemed to make sense to them. For someone with the high volume of calls that they have who are looking to send as many as possible over limited bandwidth, that probably does make sense. However, in this era of more and more available bandwidth, I've seen many people, especially on the SMB side, less concerned about conserving bandwidth and just using G.711.)

SIPconnect

I was pleased to hear Skype's Moore mention that they are looking at specifications like the SIP Forum's SIPconnect initiative as a way to help with interoperability from premise IP-PBX's out to Skype's service. Having been peripherally involved with SIPconnect, this is exactly the kind of situation that it's trying to address (interop between a premise SIP system and a SIP Service Provider). It would be great if Skype would formally get behind that initiative. (I can see certain SIP Forum people typing email as soon as they read this... ;-)

OUTBOUND *TO* SKYPE USERS

It's interesting to note what this release does NOT have - the ability to call from your IP-PBX out to a Skype user. You can call out to PSTN numbers via the SkypeOut connection... but you can't call a Skype ID. This isn't surprising, on one level, because this is a MUCH harder problem to solve. Basically every SKYPE ID would need a SIP address (a "SIP URI" in SIP-speak) that the IP-PBX could use to connect.

Several of us (myself included) have been asking for that kind of interconnection for years - and perhaps at some point we'll see that. Meanwhile, this "Skype For SIP" release gets us closer.


SOME CLOSING WORDS...

Over the years, I have written a good deal about Skype on this blog - and I have certainly been critical of Skype's closed network in the past (such as here and here). I don't like "walled gardens" in general and Skype has definitely had high walls.

That's certainly changing. I've definitely been pleased to see what they started last fall with "Skype For Asterisk" (which I wrote about here, here and here).

And now... "Skype For SIP".

If Skype implements this well, I think there is great potential. Suddenly, the millions and millions of Skype users are able to call your phone system... just via Skype. Forget dealing with long-distance or international calls. Just have someone use Skype to call your Skype ID.... ta da, it winds up on your corporate phone system via SIP. Similarly there is now the ability to easily project your presence geographically with phone numbers in different regions or countries - all through Skype's easy UI. Let alone the SIP trunking via Skype's network.

Granted, you can get the phone numbers and trunking through existing SIP service providers today. Skype just makes this a bit easier because they have the existing programs, user interfaces, etc.

Skype For SIP isn't perfect - it still doesn't get me the outbound calling to Skype users that I want - but it's definitely a step in the right direction as we continue building the interconnect between all these IP communications systems. (See my rant here and my Park Bench Manifesto presentation out at eComm)

Kudos to the folks at Skype for taking this step. For lowering their walls a bit more and letting others connect in. I definitely will look forward to seeing what evolves out of this.

Meanwhile, of course, I'll be heading over to skypeforsip.com and signing up for the beta program. :-)


UPDATE #1: (a few minutes after posting) Two items:

UPDATE #2: (6 hours later)

  • Moshe Maeir disputes my characterization of G.729 and says most of his customers use it.

  • Phil Wolff over at Skype Journal came out with his contrarian piece: Skype For SIP: Big Money, Skypeless, Brand Destroyer that argues that Skype For SIP is a negative thing for Skype. I disagree with a number of Phil's points (and would note that some of his info relates to the SFS *beta* versus the full announced release (such as the 1 Skype ID only limitation), but it's definitely worth a read.

  • Phil does make the point, reinforced in the SFS beta application, that during the beta you can only have one Skype ID associated with your Skype connection. Per the beta application, it also needs to be a "temporary" one, i.e. you may not be able to continue using that after the end of the beta period. Chris Moore at Skype indicates that this has to do with different Terms of Service and EULAs for business users right now versus regular Skype users and they are looking to rationalize all of that before the full product launch.

UPDATE #3:

UPDATE #4:

  • Rich Tehrani thinks that Skype should just call a spade a spade and refer to SFS as "SIP trunking", which is what we in the industry would certainly call the PSTN connectivity side of what Skype is offering. I think, though, that as much as we use that term, it is not widely used outside of our own industry. So as Skype seems to promote itself to the larger business audience, it makes sense to speak of it more openly as "placing calls", etc.

  • Irv Shapiro over at IfByPhone weighs in on "Why Skype for Asterisk is more important than Skype for SIP". Irv believes because SFA lets out make outbound calls TO Skype users it is ultimately more important than SFS. I agree with Irv that SFA definitely has advantage and value because of this.

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IETF 74 starts next week in San Francisco...

ietflogo.jpgThe 74th meeting of the Internet Engineering Task Force (IETF) starts Monday morning out in San Francisco. As usual there is a packed agenda with a lot of great discussions going on. This one is particularly interesting for those of us involved in the "Real-time Applications and Infrastructure (RAI)" area - which is all the various working groups related to SIP and other real-time communications protocols - as there are some proposals moving forward to rather fundamentally restructure the ways in which SIP-related work moves through the IETF. I expect there will be many involved conversations going on out there next week.

As much as I would like to be there, I won't be physically out at IETF 74. It's not my new role at Voxeo keeping me away, but rather this... oh... wee minor little detail that my wife is now five weeks from giving birth to our second child! :-) At this stage of things I'm severely limiting my travel - and flights across the country are definitely out.

Instead I'll be participating remotely, listening to the audio streams and joining in the Jabber chat rooms. Probably writing about some of it over on the "Speaking of Standards" blog I write in from time to time. The great thing with IETF meetings is that you can participate remotely (albeit obviously not to the same level of effectiveness as being in the room).

Lots of good stuff going on...


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The Park Bench Manifesto - text coming soon, video and slides now up

This week out at the Emerging Communications Conference in San Francisco, I gave a 10-minute talk called "The Park Bench Manifesto: Why We Want To Kill Off The PSTN". In the talk, I mentioned that the text would be available here soon... And it will be.

In the meantime though, I have put up both the video and the 54-slide deck over on <a href="http://blogs.Voxeo.com/ett/">blogs.voxeo.com/ett/</a>

More soon..... (need to fly home...)