Disruptive Telephony

What devices support the Opus audio codec? What softphones? hardphones? call servers? Obviously given that Opus is the "mandatory to implement" audio codec for WebRTC, it will be in many web browsers... but what other I was asked this question by a colleague recently and when I couldn't easily find a list on the Opus codec web site, I turned to the VUC community inside of Google+ and posted there. The great folks there naturally were a huge help, and quickly came up with this list:

• Web browsers supporting WebRTC:
  • Google Chrome
  • Mozilla Firefox
• Softphones:
  • Blink
  • Counterpath Bria
  • Jitsi
  • Linphone
  • Mumble
  • (Maybe Skype? They talked about it.)
• "Hard" IP phones:
  • AudioCodes 400HD series
• Mobile applications:
  • Acrobits (iOS)
  • Counterpath Bria (iOS)
  • csipsimple (Android)
• Switches / Call Servers / IP-PBXs:
  • FreeSWITCH

What other devices or software supports the Opus codec? (Or what other lists are out there listing devices supporting the Opus codec?) Please do let me know either by comments here or on social media.

Thanks!

P.S. If you don't understand WHY the Opus codec matters so much, please read my earlier post on this topic.

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Today's VoIP Users Conference (VUC) call at 12:00 noon US Eastern should be quite an interesting one. Tim Panton from Voxeo Labs and Tropo will be joining the call to talk about his experience setting up a mobile network in the middle of the desert for this year's Burning Man event.

Tim recently described the experience in a guest post at TechCrunch: "What We Learned Running A Mobile Network At Burning Man" and on the VUC call will talk more about what he did. The FAQ from the Papa Legba camp at Burning Man makes for quite an interesting read. I'm looking forward to hearing more from Tim... and the call is open for anyone to join in.

You can join the live call via SIP, Skype or the regular old PSTN. There is also an IRC backchannel that gets heavy usage during the call. It will be recorded so you can always listen later.

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Today at noon US Eastern on the VoIP Users Conference (VUC) Call for Friday, April 27th, the group will discuss the brand new "FreeSWITCH Cookbook"[1] published by PACKT Publishing. The four authors of the book, who are also leaders of the FreeSWITCH project, will apparently be joining the call.

While Asterisk generally gets most of the "open source VoIP" buzz, the folks at the FreeSWITCH project have been working away on their own solution. As they will say, FreeSWITCH performs a different role than Asterisk and is used in different contexts.

FreeSWITCH has become quite a powerful platform and I'm looking forward to learning more about what is going on with the project right now.

You can join the live call via SIP, Skype or the regular old PSTN. There is also an IRC backchannel that gets heavy usage during the call. It will be recorded so you can always listen later.

As noted on the VUC page for today's call, the show will also be simulcast in video using Google+ video and YouTube. If you are interested in joining the video side of the call, please follow the instructions on the page.

[1] In full disclosure, this is an affiliate link with Amazon and if you actually purchase the book I receive a tiny amount of money. If you think this influences what I write here, you clearly haven't been reading my site. :-)

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Heading to Burning Man this coming week? Would you like to use your mobile phone to connect up with others on the playa in Black Rock City?

If so, check out this video from Chris Pirillo about the work being done by a team of folks to supply local cell phone coverage... the vans with satellite and cell hookups are already enroute... it uses software from OpenBTS and Tropo.com to let burners leave each other voice messages, exchange SMS messages and more. Here's the video:

And here are some blog posts that provide more information:

• The Long and Winding Road to Burning Man
  
  
• Tropo + OpenBTS + Burning man = Awesome
  
  
• Voice Board and Group SMS for Burning Man
  
  
• Papa Legba FAQ (about the deployment at Burning Man and what you need to do to participate)

I'm not personally going to be at Burning Man, but this does sound very cool!

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I've long been a fan of the work that Jay Phillips did to create the Adhearsion open source telephony framework and so I was delighted to read today of news of its future. To give some context, Jay first created Adhearsion a number of years back because he was frustrated with how hard it was to create dial plans with the open source Asterisk PBX. So Jay went off and created a framework where a programmer could use the Ruby programming language to very simply create voice applications.

Jay went on to team up with Jason Goecke to further develop Adhearsion and then last year Jay and Jason joined Voxeo (my employer) to create Voxeo Labs out in San Francisco. While Jay has since moved on, Jason continues to move Adhearsion forward and announced today that Adhearsion has a new home on Github ... and hinted at much greater plans in store. I'm looking forward to seeing what all they may be...

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Long-time readers will know that I have been intrigued for a long time with what we now call "cloud computing" (and have written about it and spoken about it) and also continue to find the world of open source telephony interesting.

So naturally when I'm pointed to a step-by-step tutorial about running Asterisk in Amazon's EC2 cloud, I'm interested. :-) It's a nicely done tutorial and I look forward to seeing what people will do with it. (Unlike Mark Headd, who pointed to the tutorial in a tweet, I won't be trying it out this weekend, but I will be doing so at some point soon.)

Cool stuff...

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Technorati Tags: amazon, ec2, asterisk, opensource, cloud, cloudcomputing

Jay Phillips is frustrated. He passionately wants to see open source telephony enjoy success all around the world. Yet right now, when people think "open source telephony", they almost always think of Asterisk... and Jay sees too many challenges for developers embracing Asterisk. Jay, the creator of the Adhearsion telephony framework for Ruby, has spoken about this at recent conferences and pulled together his thoughts in a lengthy post earlier this week entitled "What We're Not Admitting about Asterisk".

Jay argues that Asterisk has not crossed the proverbial chasm for developers and outlines some of the issues he sees.

What is perhaps most interesting about Jay's post is the equally lengthy response by Asterisk creator Mark Spencer. Mark responds to Jay's various points and in doing so provides some good insight into his views on Asterisk's connections to developers, APIs, etc., as well as the differences between the markets that Digium, the company, goes after versus the "market" of Asterisk, the raw telephony platform.

Both Jay's article and Mark's response are definitely worth reading. I'm friends now with both of them and they both bring immense passion and energy to the world of open source telephony. Ultimately they both make the point that we need better tools for developers to create voice applications. This kind of dialogue is great and will only result in better tools in the end. Please do check out the posts.

P.S. Thomas Howe has also weighed in with a post saying this is a clash in world views, which also makes for good reading. I agree with Thomas that we are in a transition into a world of "web-as-a-platform"... basically a transition "into the cloud"... and so we do need "web-centric" interfaces and APIs. But I disagree with Thomas that Asterisk is "tired". To me, Asterisk is just... well... "plumbing". Asterisk is a telephony platform, as is FreeSWITCH... as is Yate... as are all the commercial IP-PBXs. Asterisk is an open source component of the rewiring of our communication infrastructure that we have underway right now. I think anyone, including Mark, would agree that Asterisk has technical challenges it needs to overcome but I, for one, am not ready to write it off yet.

Technorati Tags: asterisk, open source, voip, telephony, jay phillips, mark spencer, adhearsion, apis, voice, applications

This afternoon I'll be heading to the airport to fly out to Chicago to be part of ClueCon this week. Haven't heard of ClueCon before? Here's the quick summary:

ClueCon - is an annual 3-Day Telephony User and Developer Conference bringing together the entire spectrum of Telephony from TDM circuits to VoIP and everything in between. The presentations and discussions will cover several open source telephony applications such as Asterisk/Callweaver, Kamailio (formerly OpenSER), Bayonne, YATE and FreeSWITCH.

Billed as the "Telephony Developer Conference" it primarily focuses on the whole world of open source telephony.

I'll be there as part of two panels. First, tomorrow I'll be joining fellow VoIP bloggers Andy Abramson and Thomas Howe on a "VoIP Roundtable" to talk about current industry themes and trends. Then on Thursday I'll be part of a "VoIP Security Roundtable" talking about... gee... can you guess?

It should be a fun event... I'm looking forward to catching up with Andy, Thomas, Moshe Yudkowsky, Jon Todd and several others. There are also some folks on the schedule with whom I have corresponded but never physically me, so that will be nice as well. If any of you reading this will also be there, please do feel free to drop me a note so that we can connect.

Technorati Tags: telephony, cluecon, cluecon2008, voip, voxeo, open source, opensource, dan york, danyork, conferences, chicago, applications, telecommunications, developers

Nortel and "open source telephony"? Huh?

That was admittedly my thought when I received the list of who was going to be on the panel I moderated last week at VoiceCon on open source telephony. The other two panelists were obvious choices: Bill Miller was from Digium (makers of Asterisk) and Raza was from 3Com who have recently announced that they would be reselling a version of Digium's Asterisk Business Edition. Both Bill and Raza made sense to me. But Tony Pereira of Nortel? Nortel does not leap out at me as a company working with open source telephony - what in the world are they doing with it, I wondered?

It turns out that the answer is... "quite a bit!"

As Tony Pereira outlined in our panel as well as in conversations afterwards, Nortel is in the process of launching their "Software Communications Server 500" (SCS 500) targeted at small businesses and built using open source telephony software!

Interestingly, though, it does NOT use Asterisk.

Instead Nortel is using the "other" major player in open source telephony, the "sipXecs" product from SIPfoundry.org. (Previously called "sipX" but renamed "sipXecs" about a year ago.) I've not written all that much about sipX here but it certainly has been a product I've known of over the years. It started out as a PBX product from Pingtel which they then released as an open source version ("sipX" and now "sipXecs") and also had a commercial version called "SIPxchange". sipX garnered perhaps its most attention back in October 2006 when it was announced that Amazon.com would be using it for their internal phone systems (see the links on the SIPFoundry.org site). At a fundamental level, sipX provides similar functionality to Asterisk but where Asterisk is focused on being a "platform" for telephony that can work with a wide range of protocols, sipX is focused exclusively on SIP and also provides an extensive GUI management tool. (Pingtel provides a (obviously biased) comparison of sipXecs vs Asterisk on their wiki.)

From what I learned at our panel, Nortel is essentially creating their own supported version of "sipXecs" that they will sell as the "SCS 500". It will have full commercial support from Nortel. Target market will apparently be "small" businesses. No info really available on Nortel's site yet, although glimpses are visible through support documents (such as here and here(although this appears to be about an earlier 1.0 version last year)).

All in all it's to me a fascinating move by the folks at Nortel and I look forward to learning more about the SCS500 product over the next weeks and months as they launch it. It's a rather nice boost for the whole world of "open source telephony", too, to have Nortel making this move as well.

Technorati Tags: nortel, open source, asterisk, sipx, sipxecs, pingtel, voip, sip

I'm down in Orlando this week for VoiceCon Orlando and will be part of three sessions. Tomorrow, I'm moderating a panel at 8am on VoIP security and on Thursday I'm moderating a panel on open source telephony. On Wednesday, I'll be part of a keynote panel with Irwin Lazar on "Social networking and enterprise communication", which should be quite fun. I'll include below the full descriptions of the various sessions. If you are attending VoiceCon and want to connect, please do contact me.

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Session Title: Top VOIP Security Threats
Date: 3/18/2008
Time: 8:00 AM
Room: Osceola B
Session Description: There's been a lot of concern about voice over IP security, but have there been many actual exploits? This session will inform you about the state of VOIP security. You'll learn about generalized IP attacks that have affected IP telephony systems deployed on IP networks, and you'll also find out what VOIP-specific attacks have actually been observed "in the wild"--and what to expect in the future.
KEY QUESTIONS: * What are the most serious voice-oriented attacks that are actually being carried out? What potential attacks haven't occurred yet but probably will before long? * How do you protect your VOIP systems against these attacks? * What types of equipment and technologies must you implement to stop voice-oriented attacks? * What specific kinds of damage can these attacks cause?
Moderator(s): Dan York - Dir of Emerging Comm Tech - Voxeo
Panelist(s): Sachin Joglekar - Vulnerability Research Lead - Sipera Systems
David Endler - Director of Security Research - TippingPoint
Mark Collier - CTO - SecureLogix

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Session Title: Open Source for Enterprise Voice: How Much, How Soon?
Date: 3/20/2008
Time: 11:45 AM
Room: Sun C
Session Description: Open source PBXs are gaining a higher profile: Asterisk and other open-source PBX software packages continue to gain acceptance, and some traditional PBX vendors have implemented open source code for their products. But these efforts still aim mainly at smaller implementations. In this session, you'll learn why open source PBX software has growing appeal, and whether it will appeal to larger customers as the market progresses.
KEY QUESTIONS: * What level of market share and acceptance has open source PBX software attained? What is expected? * Which products use open source PBX software? * What are the most compelling reasons for choosing open source PBX software? What are the greatest areas of concern in making this choice? * What are the technical challenges of an open-source PBX deployment, and how are these overcome? * What are some real-world customer experiences with open source PBX software?
Moderator(s): Dan York - Dir of Emerging Comm Tech - Voxeo
Speaker(s): M Raza - Product Management - 3Com
Bill Miller - VP, Prod Mgt & Mktg - Digium
Tony Pereira - Business Leader Business Communications - Nortel

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Session Title: Social Networking Meets Enterprise Communications
Date: 3/19/2008
Time: 10:30 AM
Room: Osceola C
Session Description: It?s no secret that world of enterprise communications is undergoing a transformation; IP Telephony and Unified Communications are changing the nature of the game. Now new forms of interaction, which began in the consumer/personal communications market -- blogs, wikis and online services like Facebook ? are migrating into the enterprise. Where do these social networking systems ? and mindset ? fit into the enterprise communications landscape? Join us for a discussion about what?s real today and what?s likely to happen in the future.
Panelist(s):
Dan York - Dir of Emerging Comm Tech - Voxeo
Irwin Lazar - Principal Analyst & Program Director, Collaboration & Convergence - Nemertes Research

Technorati Tags: voicecon, voiceconorlando, asterisk, opensource, socialnetworking, communications, danyork, irwinlazar