Build your own custom phone applets with just a little bit of PHP. Rebrand and resell OpenVBX to your customers.

Give every user their own phone number and personal conference line. Dial whole departments, share voicemail messages with the team. OpenVBX is for companies and collaboration.

Adam Ballai, lead engineer on the OpenVBX project and Twilio CEO Jeff Lawson are our guests.

The Ingate SIParator® is a device that connects to an existing firewall to seamlessly allow the traversal of SIP-based communications. Ingate SIParators are compatible with all existing firewalls and operating systems.

The Changing Role of Session Border Controllers:  Ingate’s SIP-capable E-SBCs protect and enable SIP networks worldwide.  E-SBCs used to just serve a primary role of security.

We will be discussing the following pooints:

- Talk about the security role E-SBCs play.  One question many people ask us is, “Why have an E-SBC in a SIP trunk deployment if the service provider is taking on security?”  There is an excellent answer to this, one that every end-user needs to know.

- Address common security mistakes we see in SIP implementations.

- Discuss how E-SBCs enable SIP trunking and other SIP-based applications, like video, citing case studies

- Talk about Unified Communications.

Quickfuse incorporates access to data, logic and branching, instant visual feedback and a free trial account.

Geons 0.4.0.
This is the first version for public release. It comprises geond (the server), the web interface, and the basis of a distributed PBX system. For more details, including links to download and installation instructions, please see this page.

Geons is a suite of software for distributed enterprise applications. Geons applications:

• Are highly scalable, supporting millions of users.
• Can be geographically distributed across thousands of sites.
• Are fault tolerant, including geographical redundancy.
• Support disconnected operation, where a site is temporarily disconnected from the rest of the network.
• Are highly customizable. All templates can be edited and properties added, changed, or removed.
• Are extensible. New templates can be added to create new applications and enhance existing ones.
• Can be integrated with other software using an open XML based protocol.
• Are secure, with full flexible access control lists (ACLs) for all objects.

A new tool has been added to SIPVicious – svcrash.py. As the name implies, it crashes something – svwar.py and svcrack.py. This tool is meant to be used by system administrators and organizations that are receiving unauthorized scans on their exposed IP PBX.

Ben joins us to give some details about the evolution of Adhearsion which has had a recent version release:

This release marks the continuation of the platform with some major enhancements and bugfixes:

• Support for Asterisk 1.6 (continued backward compatibility with Asterisk 1.4)
• Added support for ActiveLDAP
• ActiveRecord now works properly with Adhearsion components
• Daemonizing of Adhearsion no longer truncates log files
• Escape commands sent to Asterisk via AGI
• Asterisk Manager Interface (AMI) events now work when daemonized
• Various enhancements for improved Asterisk 1.6 support
• Various bugfixes

Aaron’s experience running the service is especially interesting. He started the service using a physical host, then as the service grew he migrated to Amazon’s EC2. It’s operation on EC2 was less than ideal so he migrated to Microsoft’s Azure cloud-based hosting service. My understanding is that he’s in the process of making another migration as well as launching a new service called the SIP Sorcery Switchboard.

Oh yes, SIP Sorcery is an open source project. While most users rely upon the host that Aaron provides, the source code is readily available and some user run their own local instance of the software.

Aaron will be joining the call from his home in Tasmania, which means that it’ll be in the middle of his night! Yep, he’s a guy who writes code at night. He should fit right in with the VUC regulars.

Following the recent announcement of Bria 3.0 for Linux, Counterpath are this week celebrating the launch of their Bria soft phone for the Apple iPhone, iPod Touch and iPad.

The software is already available from Apple’s iTunes store, so by Friday all you iPhone-toting Apple-philes should have it loaded and be ready to share your initial experience with the release.

Here’s a quote from that release about the tech specs:

Key features include:

•    Full SIP compliance, enabling use with any SIP-compliant server and hundreds of ITSPs that currently offer CounterPath-based services.
o    Appears as another endpoint to service providers or PBX equipment, enabling fast, cost-effective deployments.
•    Call quality that’s superior to circuit-switched wireline and wireless voice, and based on the G.711, G.729 and GSM codecs.
•    An intuitive user interface that provides one-touch access to voicemail, call history, speakerphone and other frequently used telephony features such as 4-digit dialing.
•    Support for DTMF, which lets users enter numbers to access an auto attendant.
•    Bluetooth support, enabling hands-free calling for convenience, as well as safety while driving.
•    Multi-call management options, including the ability to switch between, merge and split calls, all in ways that are already familiar to iPhone users so they can start using these features immediately.
•    Signaling and call encryption via TLS and SRTP, enabling enterprise-class security.
•    Optional customized branding available for enterprises and telephony providers.

What’s an SBC? What does it do?  So glad you asked…as that’s just what we aim to find out on Friday!  Acme has uploaded a slide presentation to follow while listening.

In the mean time, and for the very curious among you, Acme Packet has some great background on the topic amongst their online articles and presentations.

Yes, there’s homework!

Acme products: session border controllers (SBC), session-aware load balancers (SLB), multiservice security gateways (MSG) and session routing proxies (SRP)—operate Acme Packet Net-Net OS.

HD Voice Enabling Email Addresses Via a SIP VoIP Nova Server

Voice enabling the world’s two billion email addresses offers a way to end VoIP’s unhappy reliance on telephone network interconnection. Building an on-ramp to the PSTN falls far short of the promise motivating my decision to leave Bell Laboratories for VoIP pioneer Vocaltec Communications in 1996. Realizing that original promise requires revisiting the logic making telephone numbers the default address space for VoIP.

The Nova concept leverages existing open standards, devices, and off-the-shelf SIP conference servers as a starting point. There will be a need for purpose built Nova server software, devices unburdened by legacy features, and standards if the concept gains traction. The overall approach revolves around removing the proprietary bottlenecks preventing innovation. As the presently dominant and proprietary VoIP offer, Skype represents the new AOL in this context.

You can try the system at SiptoSip and read the white paper.

We’ve been testing Skype for SIP open beta today. This allows you to connect Skype to your commercial PBX, much as Skype for Asterisk allows connection between Skype and Asterisk. Listen to the recording to learn more about the experiences to  date.

In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we’ll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

If you’re into SIP technology, you’ll want to hear this discussion about who should protect people from SIP “CallerID stuffing” among Ward Mundy, Fred Posner (VoIP Tech Chat), Tim Panton, Karl Fife, Leif Madsen and the rest of the great gang of VoIP regulars. This is why you need to join us LIVE every Friday!

Programmers differ enormously over who should filter incoming data and where. There is no right answer, although the main point is to protect your users against whatever possible attacks might come through your system or pbx.

Dan’s blog (perhaps we should say one of his many blogs) is Disruptive Telephony. Dan can be found on Twitter as @DanYork and he’s on Linkedin, Facebook, etc. Finding those links is left as an exercise for the enthusiastic student.

While I’m not sure what his current role there is, Dan is a part of another effort you should know about, the VoIP Security Alliance aka VOIPSA where he writes on the VOIPSA blog.

He will give an overview of the KONNECT Business Phones and explore the tight feature integration with Asterisk-based PBX systems.

Lastly, he will present the intriguing KONNECT Office Phone System, a multi-site office phone system that requires no PBX or hosted service.

Please hand your coat & hat to the young people at the door on your way in. It’s so much fun to have interns!

And please remember to tip the host…ok, not really. I shall try to remain upright for the hour.

It’s almost exactly a year since we last discussed AskoziaPBX. Earlier this week they celebrated the release of v2.0 after many months of work to migrate from FreeBSD to a Linux core. Michael will  be on-hand to discuss their recent efforts, the state of the software, growth of the developer team, project road map, etc.

Media5 was eager to accept our help in testing the g722 app, and I’m happy to say it works well. The newest version with the features we’re testing now should be out in a week or two. I hope to use the app from China (but who knows how that wiill work out?)

Among the participants Tim @steely_glint, Neil from IdeaSIP, Karl @karlfife, Michael @mjgraves and Graves on SoHo VoIP, David Frankel from ZipDX who furnish our great wideband conference facilities, Dave @DaveMichels from Pin Drop Soup, Pascal from Media5.

Welcome to a very old friend of mine, Bob from Future Video. The VUC encourages new people to be a part of our weekly talks. Please join us so you can chime in, ask questions and generally have a great time.

Bill Miller is a former Vice President of Product Development at Digium. Bill was also a senior product manager at 3COM and has decades of network experience. He’s worked in VoIP since 1996. I’m sure he’s got some great stories to share!

Over 25 years in voice and data networking, security and network management. Extensive product management and marketing background, Bill has held successful senior level positions at fortune 500 and multinational companies and with early stage start-ups in marketing, sales, and strategic alliances.

IRC LInks for 2010-04-23

Amazon EC2 attacks: just a PR damage control post or two, no real action.

Media5-fone: Next Friday, Pascal Dore of Media5 Corp will tell us all about the app I’ve been testing on the iPod Touch. The big feature to date: g722 wideband for use with ZipDX.

The system consists of OpenSource software and (largely) commodity hardware. The project uses OpenBTS, Asterisk and USRP. Tim will try to describe (with help from OpenBTS’s David Burgess) the technical, regulatory and logistical challenges that we faced before we could make the first call. See also David Burgess’ presentation at eComm 2009.

David Burgess, last minute tweaks

VUC regular contributor Tim Panton has been a software developer and consultant at Westhawk Ltd more than 25 years, working on a diverse range of projects, from chemical plant simulation to tourism web sites. He is also a contributor to open source projects, in particular GJTAPI a framework for implementing JTAPI (The Java Telephony API) and Westhawk’s Java SNMP stack. In recent years Tim has been predominantly involved in Asterisk development and implementation, working closely with key industry players he has been championing innovative integration of voice technologies . Tim is also the technical director of PhoneFromHere.com Ltd.

LINKS from the discussion.

Complaints of rampant SIP Brute Force Attacks coming from servers with Amazon EC2 IP Addresses cause many admins to simply drop all such traffic. Generally, SIP brute force attacks attempt to register various peer names to a system and/or attempt to guess passwords of known/guesses peers or endpoints. The object is theft of resources.

The complaints mentioned this weekend show an excessive amount of traffic; with some providers claiming 6GB of traffic dedicated to such attacks. Since we ourselves received an attack from an Amazon hosted server, we also reported and complained to the Amazon NOC/Abuse depts.

There are various techniques to assist with minimizing DDoS and Brute Force attacks, such as limiting access via the public internet, using strong passwords, not mapping extension name to peer/endpoint name, limiting simultaneous calls, and aggressively monitoring usage. Automatic blocking of abusive IP’s (fail2ban, blockhosts, etc.) can also assist with minimizing damage.

References: EC2 Abuse Report Form

VOIPSA

VUC official position: EC2 abuse costs victims time and money. Amazon is 100% accountable for what their customers do with their resources and must react swiftly to complaints.

VUC 60 second rant: This week saw a new feature rolled out, the Voipusers One Minute Issue Talk (VOMIT) where all listeners are encouraged to phone in their VoIP-related rants. Call and leave yours at (518) VUC VOIP or (518) 882-8647.

Follow  @voipusers on Twitter.

Teleku is a new cloud-based telecom service that allows Web developers to build and host phone applications that answer inbound calls and initiate outbound calls, interact with Web applications, and send/receive SMS text messages! Using simple XML or JSON responses in your HTTP Web pages, you can deliver sophisticated interactive voice response (IVR) telephony applications in as minutes!

Freedom Fone

Steve Song told me about Alberto Escudero-Pascual and this project. We look forward to welcoming him to the VUC.

This recent demo explains more about Freedom Fone.

Also; some talk about DTMF (aka “Touch Tone”) from the VUC regulars: how it works, why it’s limited to 16 keys, although phones are usually only 12.

SAM Reports is aimed at small businesses using Asterisk. I built it to solve the pain that I had : provide my clients, small businesses, with a reporting solution without the need to make any changes to their Asterisk boxes. And also to give them reporting that’s easy for them to use. I wanted to be able to offer it to both my call-center clients and those without, therefore the application handles both CDRs and queues. SAM Reports enables the end users of Asterisk solutions to have rich and powerful reports without the hassle of installing additional software.

Download audio file (TS-340614.mp3)

Guests: Daniel-Constantin Mierla, Andrei Pelinescu-Onciul and Alex Balashov (Evariste Systems)

Discussion of why you would need Kamailio, who uses it, what is does.

Kamailio 3.0.0, the first release that combines the features of Kamailio (OpenSER) and SIP Express Router (SER) in the same SIP server instance, and the vision of future development within SIP Router project.

Kamailio 3.0.0:

- new features

- how to use it to deal with latest challenges in communications: SIP instant messaging, presence, and social networking

SIP Router

- project achievements since its start in 2008

- future development

Links from the IRC  #vuc channel

Sjur Usken and Sandro Gauci have been working together doing research on VoIP security attacks. They recently presented some of their work at Hackcon, a security conference in Norway. In this discussion they’ll be talking about a number of realistic VoIP attacks and what’s being exploited by fraudsters for profit.

Sjur is a telecom consultant in Greenfield Consulting AS in Norway. He has been working with VoIP since 2002 and helping companies migrate to an all IP world.

Sandro is a security researcher and consultant based in the small island of Malta. He is the author of VoIP security tools SIPVicious, VOIPPACK and VOIPSCANNER.com. See http://enablesecurity.com/

A fight to the finish with VUC’s resident VoIP skeptic Dave Michels and Mike Oeth, CEO of  Junction Networks (OnSIP.com) on choosing between hosted and on-premises pbx.

What are the arguments for and against a hosted pbx for your business. Are they different for small and larger business pbx choices?

Dave and Mike fight it out with the help of referee Michael Graves and the roar of the crowd.

An introduction to OpenSIPS 2.0 new design and OpenSIPS Software Foundation. Our guests will be:

Adrian Georgescu (CEO of AG Projects who make Blink)
Bogdan Iancu (CEO of Voice System and founder of OpenSIPS project)
Flavio Goncalves (CEO of V.OFFICE and author of “Building Telephony Systems with OpenSIPS 1.6”

OpenSIPS, its evolution, the new design of 2.0, the effect of new design on the community, availability of more information, the foundation and open source aspect of OpenSIPS.

IRC Links

Tim Panton reports in via satellite from Nuie where they were installing an open mobile phone system.
Voipathon plans world domination with a 24 hour live Internet conference.
Randal L. Schwartz chats with us about language choices, perl, python, FORTH, PHP, Squeak Smalltalk.
“Violent” arguments about port knocking and how to protect SSH by using less obvious ports, etc.

Project DiaStar represents the work of dedicated team of engineering and marketing professionals that have been tasked with making certain portions of the Dialogic product portfolio accessible to the open source community.  This includes creating new open source projects, providing support and documentation for existing open source projects that utilize Dialogic components, working with other companies in the open source marketplace to get Dialogic products certified within their ecosystems, and promoting and sponsoring events and Dialogic partners that are focused on this growing space.

IRC Links Feb 26, 2010

Karl Fife struts his Jabra stuff. The group talked about cordless headsets, wideband and performance of same.

From the thread of the same name on the asterisk-users mailing list, why is the Morse code channel still included in Asterisk, is Morse ever used? List to this riveting discussion to learn about the rivets that hold these things together.

Tropo is a cloud communications platform that makes it easy for you to quickly add voice, instant messaging (IM), and SMS to your applications, using the programming languages and tools you already know using a web services API and JSON.

On this call we talked about the newest features in Tropo and how to get started with telephony apps in the cloud without adding new infrastructure. In fact, at least one person created a developer account and wrote a short application!

Links mentioned

[6:05pm] Ze^3k: Speaking: Jason Goecke (Voxeo) about Tropo
[6:05pm] Ze^3k: Tropo.com
[6:29pm] Ze^3k: Hosted app docs
[6:30pm] danyork: there’s also the API
[6:38pm] Ze^3k: Speaking: Ward Mundy, NerdVittles.com
[6:39pm] jasongoecke: JimCifarelli yes: on tropo-webapi-ruby
[6:39pm] jasongoecke: Code here (Ruby)
[6:55pm] jasongoecke: Other grammars Tropo supports beyond Simple Grammar are GRXML and JSGF
[6:59pm] danyork: MadManMarkAu: here’s a screenshot
[7:24pm] jasongoecke: Speech Synthesis
[7:25pm] jasongoecke: madmimi.com
[7:26pm] jasongoecke: getvocal.com
[7:36pm] jasongoecke:
[7:40pm] Ze^3k: Next VUC session
[7:58pm] Ze^3k: Speaking Michael Graves: Michael’s Blog
[7:59pm] jasongoecke: Thanks again everyone, you may follow us on Twitter too:
@jsgoecke & @tropo and the Tropo blog

Full transcription of the discussion on the next page

Jason Goecke, Voxeo:

What we are doing with Tropo is we launched it back in March at the last ECOM in San Francisco bay area and what it really is, is a platform that Voxeo has created using the same infrastructure and capabilities.  But really extending it out to the new class of developers out there that don’t want to deal with heavy standards and really just want a simple API to start developing applications.  We have seen a trend out there where things are moving from things like SOAP XML to REST, JSON, these types of simplifications and Tropo is really a response to that for the real time communication and voice world.  So what we have been doing is creating a platform where you may easily come in, develop an application in a scripting language of your choice such as Java script, Groovie, PHP, Python, Ruby so giving that choice to users once again to give them a comfort in whatever language that they have been developing in, and be able to use a simple but powerful API to actually drive telephony applications.  Those applications include things like outbound notifications, inbound self service, the ability to set up conference calls, all of these capabilities.  So what is interesting is I was actually at a Ruby meet up the other night here in the San Francisco Bay area and when talking to the Web 2.0 crowd and people who were used to doing web development, when we showed them how you can write a five line script in Ruby or PHP, and your phone rings or it answers and says something, they are completely blown away and they start to plodding and think it’s the best thing they had seen in months.  So it gets developers quite excited when they see that simplicity and can mix it with their own applications right off the bat.  That’s really the goal of Tropo.  And I think we spoke probably last fall or late summer timeframe and what we had at that time was really a platform that was voice capability where we had speech recognition, text to speech, you could dial in over a VOIP, you could used Skype SIP or get a phone number and dial in to your applications as well as dial out to the world as well.  Now what we have done and what we focused on since that point is the trend what we are seeing that users and customers and folks out there don’t want to be trapped in just using voice to talk to an organization.  We are seeing a big trend towards wanting to be able to use multimodal applications whereby you can send in instant message, you can talk over Twitter to an organization, or you may use SMS and one of the acquisitions that Voxeo made in 2009 which was quite key is a platform called IMF8 which really is a single API that unifies all of those into writing one application that you may then deal with all these different disparate channels and mediums for dealing with your users and with that we have actually added that into the Tropo platform and now using that same simple API you may go ahead and develop those applications using the same back in the same script even, and now you are actually talking to people on any medium that they choose.

I should have said right away we did put this in the RC, we are talking about TROPO.com and you should go take a look at that while we are talking here to Jason and eventually Dan if he wants to chime in.  How do people get started with this?  What’s the best way, I am sure there is some way, there is probably some apps you can get started with and so on.  What would be the best procedure, do you need to register, you need to go to the site and join, it looks like a join now thing here, what about that?

Yeah.  What we do and this goes back to Voxeo 10 years all the way through today is we believe that to develop on the platform should be entirely free and the same experience you would have once you went production.  So that means you may sign up for an account as a developer, it is free, you may go ahead and write as many applications as you want, you may get phone numbers associated to your account as well, and just start developing and that includes all the capabilities of the voice which is text to speech and speech recognition, all of the SMS capabilities, instant messaging etc, and it is really a simple, it takes 30 seconds to sign up, no credit card required or anything like that, and you are off and running.

You can I could develop an app right now, click on join now, and we are done, five lines later, right.

Absolutely.  And of course we have a lot of documentation out there, we are always working to improve that documentation, we get feedback from our developers and users about how we can make things even more clear there, and we have probably 30 sample apps out there today in a whole host of different languages that people can plug into their account and immediately have something up and running.  That includes things from checking the weather with Yahoo to getting Bay area rapid transit information to dealing with NFL highlights or things like this.  So there is a whole myriad of applications out there that you may instantly get started with as well.

Is it common in IRC about integration with Skype?  Is that mentioned in some place?

Yeah, so what we have is when you sign up for an account you actually get three different ways by default that you can dial into an application, and that includes a SIP URI so you can actually dial in from any SIP cell phone, it includes INAM, so we have INAM support in there as well, and you get an INAM phone number, and then lastly you actually have a Skype +999 number that you can dial to get into the platform as well.  And all of that in for easy development as well.

And we should be clear that could be INAM too depending upon how you say it?  INAM

I am sorry.  I am used to INAM, and so yes the INAM support, and then you also have the ability once you set up that application to also go ahead and add a phone number or DID from various different points in the US.  We don’t have international support for the developer account yet, we are working on that, but in terms of getting phone numbers into that as well, you are up and running very quickly.  And you can have really as many phone numbers as you want on the developer account as well.

How would we able to compare Tropo with Twilio who have also been our guest by the way and are very gracious people?

Yes, I was actually on that call as well.  So it’s actually Tropo for troposphere, we actually had this debate once and for all the other day internally is it Tropo or Trapo.  It is Tropo.  So, what we have done as well,  I want to go into some of the new features and come back to that question, also.  We have actually as part of the new capabilities we have added since the last time we talked is the ability to have a remote API as well which we call the Tropo web API.  So we had historically is what we call Tropo scripting and that’s where you write a script, you post it to your account, and you are off and running, similar to a Google app engine where you can actually just run your script in the cloud which happens to be our cloud and you are off and running.  One of the downsides of that is we don’t have every single library and every single language you may want running in our cloud, so the alternative is we have come up with a REST JSON API and that’s where JSON comes in and go ahead and be able to run your applications remotely and have access that seems simple API in our cloud. So that’s part of what we wanted to do, it does give a lot of developer choice in terms of different programming models, different ways to access us etc.  So that has been one of the big things we got and that has been out there now for several months as well.  Now how we compare to Twilio, they have the ability to do a XML API over HTTP and they have had that now for well over a year and what it comes down to is Voxeo has been doing XML APIs with VXML, CCXML, Cove XML now for about 10 years.  And what we have seen is that the developer community is looking for greater simplicity, we didn’t think the world needed yet another XML API, we feel that’s well covered at this point, and hence why we focused on the Tropo scripting APP engine style, we focused on the JSON capability because myself as a programmer I know it is a lot easier to deal with JSON parsing than it is with XML, and that is the trend we are seeing out there with APIs around Yahoo and Google and these guys.  And further we are using the same backend infrastructure that powers Voxeo which is arguably the largest if not one of the largest cloud platforms for telephony and real time communications which means Tropo gets access to that industrial strength capability which includes things like full speech recognition, and includes the full multimodal capability and really on a cloud that is optimized for running telephony applications.  So I think most of the people on this call would know that handling voice optimization in the cloud is quite a bit different than handling web optimization with the likes of Amazon and the other cloud providers and what Tropo doing is that the table being powered by Voxeo, is really that voice optimized cloud backend, we have multi carriers that we deal with, so we don’t have outages, we have multiple data centers, our SMS capability isn’t through one vendor, it’s actually through a myriad of vendors, so we really bring that professional Voxeo background behind us that gives us all those capabilities that some of the other players out there don’t have.

I am trying to encourage people to get in on audio and ask you questions.  I think Jim’s question got asked already, but he has a crying baby and can’t, anyone who does not have a crying baby or barking dog, let us know, I guess the IRC if you have questions?  This might be the time to talk about, this is always a little hard to do but if there is anyone or any applications you can mention, would I mention the name for example of what people are doing?

Yes, I can actually give you a couple of examples.  One is a company called Awayfind.  Awayfind is a organization that’s focused on creating let’s say, alerts out of your email and your different channels that may be coming to you, and then escalating those based on certain business rules that you set up.  So the whole idea is I got information overload, I may be away from my desk, I have got my phone with me, I want to actually set some business rules in terms of what gets to me on which channel where based on status or on different pieces of information, and I want to go ahead and push that out to the user, and they are actually using Tropo to do that, to do the outbound notifications.  So, if I have an email that comes in that’s important from someone and I am sitting in a meeting, I want to get an SMS, so I want get a phone call, and I want that message to be played back to me.  So that’s an example of the outbound alert notifications.  Another one that we have recently been able to talk about because they get all their IP filed away, yes Orange Labs out of San Francisco is actually developing an application that they are calling Status, which is really an all encompassing social network mobile application where you can do things like set your status, it then gets communicated as text to speech in your voice mail for example, so I just got an airplane, I want people to know I am going to be landing in Denver in two hours, I can quickly tap in a status update into my Android or I-phone application, and I could translate it into my backend voicemail system for example, and that’s being used by Tropo.  They are also doing things around the ability to add instant messaging interactions with you and your status and publish that information, so they are actually using a lot of the Tropo features around both the inbound dialogues, the outbound capability to create a really seamless environment in terms of, if I am running in a social network I don’t want to be limited just to my Twitter client for example, I really want to be able to take in all of my channels and all the different things I may be doing and have that feedback in to what my status is.

Jason, I think I have been spelling your name wrong for the past twenty minutes now.  So forgive me everybody.  I don’t know how to pronounce it, so that’s what happens.  You mentioned SMS, and as you know I am here in Europe and SMS has always been huge here, and frankly I am kind of cheap on portable [unclear] I used SMS a lot myself, but the folks in the States, I often hear people who are little confused by, and it is 99% mobile, I don’t know if there is fixed line SMS in US, is there such a thing?

We have Google voice

Right.  But besides that, I mean does anyone like this [unclear] give you SMS possible?

Not really, but what we have done which is interesting and kind of surprises people in the US market is, for SMS today, you had to have short codes.  What we are doing on our SMS platform is, we are giving you a regular 10 digit phone number just like Google Voice would for example, and every phone number now on Tropo is SMS enabled right.  So you send and receive SMS from that same phone number that you dial into and out of your application from.  So you are seeing a trend towards, and I think it is somewhat driven by Google Voice for a lot of people here, but people are beginning to realize SMS is not mobile only but can actually be used now in conjunction with phone numbers across the board.

And what are some of the [unclear] real quick that one of the things we used SMS other than sending messages to people, one we were using Asterisk as our PBX, I used SMS in both directions to get notifications of someone who calls and didn’t leave a message for example, or the incoming I would send command, but totally insecure by the way, by sending command ‘call me back’ in Spain in a phone booth and stuff like that, via SMS, that worked beautifully, but what kinds of interesting things, we are talking about just sending some SMS, hey party, which is good, or is there other particular applications, here for example we work in the wine business and wine producers most of them have SMS alerts when the temperature of the mix of grapes from [unclear] gets to a certain point, that sends an SMS, there is industrial process as this one, what are uses there besides partying in the US? Anything interesting?

I think you hit on one of the network scene which is around what Thomas Howe likes to call the communication enabled business processes where in the particular business process it makes sense to get an alert via SMS so if you have people in the field on their mobile phones, they take particular action in whatever that business process may be.  We are also seeing applications where people want to use it for a secondary piece of authentication whereby you can make sure that actually dialing from a device that you would expect them to do so, they call in, you collect some information, you SMS them a code and then they enter that code in, so another step of verification on that as well.  I was about to finish up and say, we are seeing areas like Awayfind where you are using it for escalation of other channels coming to your SMS for immediate attention when you may not be sending at your desk for your email and things like this.

Well, its [unclear] in New York, I would like add in or two, we are seeing a lot of interest from folks around the whole multimodal, multi channel interaction with a context and with a self-service kind of application.  So it could be you want to check the status of your package, typical package tracking type of thing, and you are standing in line, so you can’t make a phone call, but you can use SMS.  So we are seeing people looking at it in a two way environment where you are interacting with a service to go and check on the status or something or to communicate with a text support center or anything like that.  So we are partly enabling not, it is really with Tropo you can do, it is a toolkit.

So what do you want to SMS enable, how do you make that happen?

Interesting, because that’s an excellent use for it and I guess they do the same mistakes, you can get your minutes via SMS.

That is a very good point because a lot of superfluous use of cells that you really didn’t want to know when you are standing in line, it is much better [music] that was inevitable.  That is Strike 3 [unclear], nice to know you have breaking news.  Breaking news has a caller who has a question.  I was just saying that SMS, that is actually good use of SMS, Carl are you with us.

Yeah, I am.  So my question has to do with more than the mechanics of how you are implementing SMS, in other words, I know that using some technologies you can send binary formatted SMS messages, for example to do something like over the air provisioning of handset or tweaking certain settings on a handset, and so I am wondering are there any capabilities for sending binary formatted messages in your framework there, and then second question is also related to more than mechanics of it, in one of our pain points right now is that we have terrestrial numbers that we want to able to relay SMS messages out to a mobile phone that can conveniently process a text message and it is difficult to find a service provider who will do it out, who will let you port your number and then also, programmatically deal with your text messages.  So I am wondering you said that you and Google Voice apparently as far as I know are the only ones that have put an indication that have support for text messaging, so my question is whether I can take a number to you and do that or are you getting your numbers from a pool where you have a partnership with someone who has an SMS and is something like that, so yeah that’s my question.

So on the first one in terms of the binary format within our interface structure and within Tropo specifically, we do have the ability we are using scripting languages etc to send out binary information.  I would have to go back and check to see if we can actually do it at the SMS gateway implementation that we have and I am pretty sure we can, but that hasn’t been something we have gone out and proactively tested as of yet, so that’s something I have to differ on that side to be 100% sure.  In terms of portability, we have the ability to port numbers into our network and then the way we are doing our SMS is we could actually SMS enable new phone numbers based on the partnerships that we have.  But it would mean porting your number into our network which has locations round that as well.

I see, okay thanks.

I was just typing in Talkshoe and for everybody listening if you are not on air or sea, it would be a really good way to join us and that would be at VUC.ME/IRC which would bring into a web, if you don’t have IRC client, it will bring into the web interface and then you can tell us if you have question or if you are not able to call but you are listening, you can ask that question, one of us will ask it for you, and while we are talking about questions, anybody else have any questions?  Type it in Talkshoe real quick, sorry, but in the IRC channel or if you called in unmute yourself and ask the question.  Anybody?

What is the pricing on the SMS functionality?

Yes, what we are priced at is 2 cents per message and that’s two US cents per message, whether it is inbound or outbound.  That’s what we are charging for the SMS side of things.  Now, remember on a developer account to get started and again testing in plain with the service, there is no charge.  We provide those free of charge.

Two cents seems almost absurd, inexpensive to me by the way.

Yes.

That’s like a pretty much of a mass raid.  Any quick question on SMS also, Quid of international SMS, that usually nobody does that?  Can you or will you some day?

Yes we do have the ability to do the international SMS as well, as part of that.  So really it’s using the same SMS network to be able to go ahead and do that, so yes.

That’s interesting.  I am not sure that Google does that, a caller is asking if it is the same rate, I don’t know would it be?

I see that, I have to go back and check on what the rates would be on that side.  I am not a 100% sure.  So I don’t want to misspeak.

Yes it is not immediately obvious.  I have been in countries where it is, and we are told it is the same rate, but can you believe what Orange says anyway.

I like Orange, Orange has developed a great application on Tropo.  So I can only say good things about them.

Somebody can, but as a customer, I can’t say good things about them, but let’s not go there right now.  In the absence of any other questions, here is Carl saying another company charges 30 cents per international SMS, now I don’t know that wouldn’t be Google Voice, I don’t think.

Well, that’s the thing, Google Voice I am able to send SMS on Google Voice to like, I have a lot of family over in Spain and I am able to communicate with them without a problem.  So I don’t even think I actually have to pay for that, if I recall.

That’s great.

And just want to mention also that, I saw some place that Sipgate was doing SMS as well, just to be fair, but I don’t recall actually seeing that.  I just heard it.

There are a lot of companies that provide SMS, Termination SMS Services and those types of things.  So there are a number of folks out there when you get into the infrastructure side of things.

And this is recent though because nobody has heard of SMS outside the Europe for a long time.  But I think it is a good thing for reasons that we have already said.  SMS is cool. Yeah.

And don’t forget, what we are also talking about is instant messaging as well.  So in that same way you may easily send out SMS back and forth.  We have the ability to do the exact same thing with the same application out to Yahoo Network, Microsoft Network, Google Talk, AIM etc and have that same type of interaction.

Now, are those priced at the two cents also?

No, right now those are not charged for.  We are looking at various pricing plans in the future where we would work rate something like 15000 messages at X, it would be nowhere near 2 cents, it might be 0.0002 cents right, something along those lines, because the infrastructure to deliver instant messaging is quite a bit different and I want to here now emphasize there is no charge on the developer account for that.

Right.  I understand.

I just want to say really glad you if you dialed on G722, it sounds really good.

Great.

On the instant messaging side too, it is [unclear] or XMPP etc, so there is all the different pieces in addition to what JSON [unclear]and also Skype IM as well, for we can support in parts of that, so we have the ability to do that and one of the interesting and compelling things about Tropo for me is that you can do this all from a single application.  So to one app going across all those different channels.  So you can write it, you can tweak it different ways if you want to for the different channels, but you don’t have, it is just one app that is able to go and work with however people want to interact with it, whether they call it, whether SMS or IM, so any inbound or outbound, neat stuff.

Yeah, and maybe I should walk through an anatomy of an application quickly to give you an idea of what that single application might look like that can handle all of those channels.  So what you have is the ability to write the script as I mentioned in five different scripting languages or if you are using our remote API and virtually anything that can do HTTP web serving.  So what you have the ability to in Tropo is a set of verbs, methods or actions that when you write a script and post it to us, they look something like you have a line that says answer, it simply says answer, and then the next one might be, say, and then you pass it a string ‘welcome to my application’, and it does a text to speech back to the user whether it be a voice call or an instant message and sends that information back to them, then you have the ability to ask and you simply say ask past a string, and then you pass it a simple grammar which says okay, this is a universe of possible responses might ask for some, maybe I am going to ask you what your favorite language is, and I can actually pass a string that says okay I am expecting for Ruby they might say, Ruby and JRuby, for Python they might say Python and Py or Jython, and that gets returned in your application when you ask them what they said and then you can go back and say, okay your favorite language is Python, that is the next line and then hang up.  Right, so you have written a very simple 4 or 5 line script with the ability to set the speech recognition capability and then that works over SMS, instant messaging or voice in equally the same way.  And you do have the ability also to do things like detect what the first message they sent was, because when you answer the phone you don’t generally look for the hello, or the information the user may first say, but in an instant message they have already said something to you.  So we have an application for example Yahoo weather where we ask for your zip code and then we go to the Yahoo web API and playback your local weather, in that one a user who may have used your service a few times, may just send you the zip code in the first message you want to be able to capture that and immediately respond to them without asking them that information.  So you have the ability to write one application for everything, you could also have the power to deal with the nuances of the different channels that you are dealing with, and to that end you can detect was it a text call, is it a voice call and what network did it come from, did it come from Yahoo, Microsoft, Gtalk, or an SMS etc.  So you can write one common application as well as deal with all those different nuances and subtleties of the different channels that a user may use.

I am guessing that you don’t quite have that ready yet.

No, that works today.  No, we don’t have [unclear] sorry I didn’t see that, no [unclear] yet but we have had people that have actually written Closure and Scala and things like that for the platform but supporting five languages is already quite that.

And to be clear on that too, to talk about the nuance there, like Jason said earlier, there are five languages that we support in our host infrastructure.  You can upload them, you can upload just script into our hosting cloud and have it run there.  Very much like the Google app engine model where you upload your Python code and add Java or whatever app into Google’s site and it goes and executes there.  So it is a hosted model where you are up and there and it is running, and then what we announced just a little bit ago was the web API where it is running on your site, and interacting with us over that REST based the JSON REST based API so you could write in whatever language you want, so I am not sure who asked this in the chat, but he mentioned, could you write it in CURL, and sure, I mean it is just a HTTP, it is a REST call over HTTP so you could write it in whatever you want to write, and Gregory, he says you can write on whatever you want, I mean you could write it in Coldfusion, [unclear], Fortran or whatever you want to do.  That’s the difference.  On the web API it is running on your server and interacting with our platform over the web API but it is written in whatever language you want, if you host in our platform, we have got five languages you can choose from, so that is the nuance there.

And the key there is what we are doing is we are providing developer a choice, a developer to get started may just want to write that five line script, post it and host it on our platform, so you need only to have to host it anywhere, and they have an application.  Or someone wants to do lots of backend integration to whatever application they have got, you can go ahead and do it that way with the Tropo web API and another thing not to forget is that Voxeo sponsors the Adhearsion project which is one of the primary frameworks that works with Asterisk and one of the tasks that I have here shortly is to actually write an Adhearsion component that will integrate by default to the Tropo cloud, so for example in Asterisk if you want to add in voice recognition you can do that by calling in a Tropo resource to actually do that and then release it and carry on within your Adhearsion in an asterisk application.  So we are not forgetting all of those implementations out there, that use Adhearsion to do these types of things as well.

I know that Carl is dying to ask to another question, before you die, go forward?

Carl:  If I write an application up in your cloud how are you trunking that system out to the network, are you doing SIP trunking over the public IP network or are you using PM trunks to the ILAC or how do are you trunking that out.  The reason I ask is because, everybody here knows me as the latency [unclear] and there is a lot of application that I built and sort of scrapped or had to re-engineer using faster TDM trunks so as not to basically create sort of a latency penalty, using that application and it is one thing to say yeah I can still talk to the person, but in our case we had calls that were going out to a sales center and basically what it was is when you are on that call you had just didn’t like the call as much, it just didn’t work as well, and [unclear] was trying to save the nuanced subtle aspects of talking with our business by not having it to be one of those calls that you don’t like for some reason because it is too slow, so anyway that is my question.

And it is a very good question, I am glad you asked it, because people that come from a web background don’t understand the subtleties between a web interaction and a real time application like SIP and RTP.  The Amazon cloud is great, it works great for the web but it is web optimized.  And people regularly see spikes of 1000 milli second latency in and out of the Amazon network which is fine if you are doing a webinar action, not so good if you are trying to carry on a professional conversation with the customer.  So that’s one of the areas that I really like to emphasize with Tropo is we are using that same infrastructure that drives one of the largest telephony clouds around which is Voxeo where all of our trunking is directly with the carriers.  We don’t do trunking over the internet, so we do have SIP trunks for those are direct SIP trunks with the carriers.  We do have TDM and convert the TDM to SIP as close to the edge in every case we can because our internal work is 100% SIP, but that is within our network.  So when we are trunking out to the broader world, we are very cognizant of latency and what you need to do to provide professional quality for real applications in real business and that’s why you won’t see us sending or calls over the broader internet in the carriers, they are all interconnected, so for example our data center in Orlando sits right on the rail road tracks and we are running fiber directly from three plus major carriers in there, so not only are we directly connecting, we are actually interconnecting with multiple carriers, because you know a carrier does go down once in a while, they do have trunking issues etc, so you have to immediately be able to route between those and that’s what we really drive home around the fact that we are a professional real-time voice platform, not just something running in someone else’s cloud.

Yeah, someone else’s garage.

Correct.

To extend on that a little bit too.  Voxeo has been around 10 years and has done a lot in the space on XML side, on voice XML and CC XML and we have literally tens of thousands of customers who are using us for all of this.  And we have a very large scale and that is brought about by having multiple carriers and, we are one of the, it is funny I am SIP guy and when I came here two years ago, it was interesting to learn that Voxeo is one of the larger consumers of SIP trunks in North America.  Like Jason said they are directly with the carriers, but it is a huge amount of capacity and so we can do things like we had a customer recently who did about 1.3 million calls over a space of about 90 minutes, and it was and for large amount of what they were trying do for, it is outbound notification type of stuff, and that is the kind of capacity that we have built into the network.  So we are also very latency concerned, because we have to be when we are providing this amount of capacity at this amount of scale, we have to be concerned about latency.

For customers that already have an existing SIP infrastructure, I guess what I am searching for, are there any cost savings to them by channeling the stuff out, be a SIP versus using regular phone numbers?

What they could do with SIP, because we are SIP cloud you can interact with us in ways that you couldn’t do in the traditional TDM environment, as for instance we have some folks who are using applications on our site for say, customer satisfaction, when in those call center, one of those would you like to be part of this interview when this is done, and what they are doing is from their call center they are interacting with us via SIP and in doing so they can send rich call headers across in SIP to our platform that can denote various different things like where the state of the application was, information about the caller etc that you just can’t get from simple called ID and so we can process it on our side, have the call interaction and send it back to that other site again through rich SIP so that they can pull back further into their system.  One of the advantages is that you have that ability to go and provide a richer context to the communication.  Now somebody won’t limit your talking SIP to our cloud over the internet if you want to, but that’s a choice.

We have some question that was, are they still going to be at the 3 cents a minute right, where it is pure SIP to SIP, for example you know large banks that have SIP connections between all of their offices, to take advantage of Tropo are they still up to 3 cents a minute rate, even if it is pure SIP traffic.

Well, the 3 cents a minute rate is for that user that comes in once to get started, starts building your application, goes out there and begins growing etc, if Bank of America and [unclear] wanted to start using it of course, you have a whole different discussion on that level.  Just like anything else, it becomes the scenario as to what is needed for that particular situation, because one of the things on Tropo is it doesn’t have the same SLAs that we have on the broader Voxeo network, it uses the same infrastructure, so it pretty much has the same operating capability, but for that 3 cents a minute, you are not going the same level of SLA and if you are a larger bank, even in if you are interconnecting with SIP you are going to want an SLA with penalties which comes at a premium.

Are you folks interested at all in giving any or capable of giving any statistics for minutes per month or anything like that?

What has been to date is Tropo has been a beta platform, we want it to really get a lot of testing, a lot of users on it, lot of developers which we have successfully done, and we will be launching production shortly where those minutes will actually become something that we start talking about regularly because the developers are using it in a whole myriad lot of ways, we are seeing peaks at certain times and troughs in others, and part of that is developers are looking to take it production before long and that’s exactly what we are focused on at the moment.

Did you mention any numbers, I did put the URL but are there phone numbers what they are most that are public…

Yes there is actually one right there on the front web site, you will see if you scroll down on Tropo.com, to the bottom you will actually see an example application you can dial SMS or IM into as well, that number is a US number, I actually put those details in the IRC as well, that you can dial into and actually hear an example application and then of course to write your own, you can get a phone number for free as a developer as well.

I don’t know probably if I asked this question or not, I am not sure it was the same question, if I got, can I port the number in?

Yes it is possible to port numbers in, right, but it takes a little more time and process to do that.

You don’t do that for like a 20 minute experiment, I mean [unclear] done your testing.

Exactly.  When you are ready to launch.

And also to the questioners, if you want to play with it, we in the docs.tropo.com  it is there in the just basic part that’s labeled there is app engine, there is a scripting part of things, there are some tutorials, VB tutorials, Java tutorial, PHP, Python, ruby and there are some samples that go through, that have all the codes, there are working samples that people can go and just create an application up on Tropo in a hosted environment and upload the code, and you just copy and paste the code in and set it up and use it right there.   So it is that easy.

And to that point, I would like to extend an invitation to everyone on this call, is next week we are actually launching a private invite only beta of some new features that we are about to launch which are quite interesting and if you go ahead and sign up for an account and then email me your account name, I would be happy for any one that is on this call to go ahead and participate in that beta.

An excellent opportunity, Jim [unclear] I don’t know if you are dialed in, you have a question and I was going to read some of these numbers into the [unclear]

Yeah, can you hear me okay?

Sure, great.

Jim:  Hi Jason, it is Jim here.  Great stuff.  I am interested in signing up for a developer account and putting together maybe some projects for potential customers.  If I sign up for a developer account and say I have put together some sample software, can I keep that account around for a few months, or what if I don’t dial in or use it for three months, do they become inactive, or I hate to tie up a phone number, that sort of thing, maybe you can comment on it.

Now we leave accounts open as long as you want to, you will have to express we access to close it and even then we probably leave it there and just change the password.  So there is no limit in terms of you coming in and signing up, getting a phone number and having that account, you can keep it open, in fact while Dan and I where speaking on some forums down at IT Expo a few weeks ago in Miami, Thomas Howe pointed out how he actually has account number 68 I think, on Voxeo and has had that since the very early days.  So there is no expiry.  We are not worried about you taking up space or a phone number because really we want to encourage developers to come, give it a try and then come back when they need to, and not put any friction in place that would keep you from making progress on the Tropo platform.

Great, thanks Jason.

Let me just state one thing here too for everybody who is listening to this.  Some people are not regulars and they may have just jumped in on this, been interested in Tropo and Voxeo, we have been in this call for almost three years now, and we feel like a community, many of us have met in person, and I just want to underline the fact that people who are guests like [unclear] on this thing had been on twice, and these are real human beings that you can contact.  Jason [unclear] as opposed to bots….what I am saying is this is important.  Let me put it this way, it is important to me and other areas of business when you meet somebody online and in fact these are guys who are interested in hearing from you.  There is no joke and we have had a lot of interesting interfaces with a lot of interesting experiences with all kinds of guests on the show.  So if Voxeo, if what they are doing on Tropo was interesting to you, you want to get in touch.  It is a kind of a privilege contact and I want to underline that because this is not some kind of an ad for these people.  These are guests as people we know in our community and they are part of that, Jason is an innovator, Dan is a very well known figure in the world of VOIP and I just have to say all that to cover the fact that I am [unclear]

And we have also been on this show, I mean I have been on here as a participant any number of times too.

We feel we know you.

You don’t know actually, you are interacting with Tropo app right now, that’s really me, I am not really here, it is artificial intelligence.

I just got an SMS advertising some kind of plan I can sign into with Tropo.  Whose else is interested in asking some questions, because he is [unclear] human being, stump the bat.

Actually I was going to mention that we could have probably worked something out to connect to something like bridging with ZipDX but it is such a lot of work to do for something that wouldn’t be obvious immediately.

Maybe after the main call, we have a little after party, we can do something there?

How do you know about the after party, guess you were not told about that?

Oh sorry.  This is the best touring test that I have heard.

We could have attached a bot to automated agent to this all, that could have listened for certain keywords or something like that.

Can I ask about speech recognition, if I missed any comments about that, then I apologize.  If you said something about that sort of shut me up, but I am curious to hear you say a little bit more about what kind of speech recognition you are doing, limited vocabulary, or not and obviously it is speaker independent I am sure, maybe you can just talk a little bit about how people are using that and what their experience is?

Absolutely.  And a key thing to point is I believe we are the only platform out there that includes speech recognition and at 3 cents a minute.  And that is huge piece of developing voice applications.  So it is something we are actually very good at.  So we have our own speech recognition engine that we use which does utterance based, grammar based, speaker independent speech recognition, we do have what we call simple grammar and once again we are after simplicity first, and then if you want sophistication, we don’t block you either.  So for setting your grammar switch or the ability to set your possible answers into what those values will mean, such as if I am going to call up and I want to check the rates on a particular airline, I might say United Airlines or UAL or American Airlines or AA and be able to return what those things mean and we have a very simple capability to do that that is Tropo.  Now having said that we also have the ability to do full GeoXml which is the grammar XL format and you can feed that into Tropo.  We also have the capability to do the JSGF which is the Java capability as well, so if you want very sophisticated speech recognition beyond simple grammar you have that capability as well.  And I would like flip that on the text to speech side, it is a really very good text to speech engine where you can just pass it string and have it say something, and you can also pass to it SSML and with SSML it is another XML format that allows you to go ahead and set inflections, set speed, set what effectively becomes emotion in the voice and some very sophisticated things.  The whole idea of Tropo is make it simple as possible to get people engaged.  And as they want to expand our applications you have a road map on every aspect of it to go forward as your application grows and you need more capabilities.  We don’t hide any of that sophistication if you want to get into it.  And one of the things that we are testing is some enhancements to our speech recognition for Tropo and that is part of the private beta that I am talking about as well.

So in your experience, if I just wanted to start simple, and I am a big fan of what you said, and I wanted to offer a few words may be numeric entry as part of idea or kind of application, is the performance pretty good on that.

Yes, I mean, I have actually been quite amazed where I am just trying to type in a quick example into IRC here of what a simple grammar might look like as well, you have the ability to say something like, okay in the first part I want what I recognize and I am asking what your favorite language is for programming, so I might have Ruby and then in parenthesis I put I want Ruby or JRuby or PHP or Python and Pie and those get recognized and return back to your application.  What was interesting at the Ruby meet up I was at in San Francisco the other night, I had put up an example to say hey Rubyrail that’s your favorite language if you put Python here, you are crazy kind of thing, get them clapping stuff like that, and one of them said hey, I see you have JRuby up there, it certainly won’t recognize that, so I was a little bit worried because I hadn’t actually tested it at that time, so I called back in over Skype said Jruby and sure enough it came back recognized Ruby and everyone was quite impressed with that.  So both in terms of performance and the ability to do recognition, it is a very strong pop form and we have a lot of customers that are using it for production applications in broader Voxeo as well.

Great thanks.

I have been trying to trap or band something here, Dan here is a well known security expert and we should probably talk more about security.  Now what can we say that’s not going to be too revealing here, everybody is worried about that.  So here we are with out application in cloud and what kind of things can we talk about to make sure that security is done right besides the fact that you are involved?  In other words if you are my [unclear] and you can tell me if it is going to work, but that’s obviously has been thought of so, we had hacking VOIP here, so what can you say about security, because I am sure there is concern and it is probably not something you would put on this site as a page, so we are going to talk about it here.

I mean it is a concern and it is something that we do talk about and you know me Randy as far as that I am certainly out talking about security in webinars and various different blogs and other sites with the VOIP security lines in other places and, security is certainly part of what we interact with and it is part of what we have gone through our testing through our own systems to look at how we make our systems as secure as they possibly can.  There is a larger issue really which comes in to when you are looking at doing pushing your applications out into the cloud, there is a natural concern about how do I trust the cloud to be there, and you are going back to sort of what Jason was talking about earlier, this is part of what Voxeo [unclear], because we have been doing this for ten years.  We built the infrastructure, we have multiple redundant data centers, we have redundant internet connections going into those data centers, we have redundant carriers and multiple capacity so that we can be able to go and do that.  On the production side of the house, on our traditional XML site, we have a 100% guaranteed SLA which is basically that we will give you money back if we are not able to give you the uptime that is 100% and we do that because we put in these layers of redundancy and availability so you can be sure that they were there and that is something we have done, and so that’s a large part of that security.  It is how do you trust that it is going to be there, that’s going to work with there, there are other elements too when you get into dealing with certain voice applications, there are lot of issues around compliance and ensuring that you are not capturing private information when you get into dealing with HIPPA and other [unclear], so we have capacity within our environment that you can not record certain pieces or rather parts like that.  So it is another step that we have taken inside there.  I think the security is like with anything, it is multiple different layers and some of it, the availability, some of it is the, what you can do in the application, how we segment the applications so that your applications can’t walk over other people’s or use other people’s capacity, it is all of those different types of things.  So I don’t if that is a precise answer but it is a big wide topic you probably talk the whole time on.  So I don’t want to take a lot of time.

You do have a credential, there is no question about that.  So it is not a problem, I wonder if you have anything to say about it?  When people sign in, the only thing I can think of that you can do that some companies do with regard to that is as far as passwords goes to port to force good usernames and good passwords that are not easily guessable.  I had a customer over here who I was helping move his whole thing from Exchange over to Google Apps and I needed his password.  His password is a phone number, it was like hello, people don’t learn that, granted it was a phone number from like 1956, but still the point is, for example on SIP.com they force credentials, they are the ones giving the credentials, and that way it is impossible to have and you can’t change them, you can tell us to generate new ones, so that would be one example of things.  I am sure you have done things that whatever you can do on your end to make things secure.

Right, we try to make it as frictionless as possible for developers to get going and do all that.  I am a back end and we are certainly monitoring stuff and certainly watching because a developer could create a application that goes off and does nasty stuff, so we have monitoring that we do to make sure that things are not going too crazy and stuff like that, so we take that very seriously.  So it is something that we are doing a lot of on the backside.

And Jason is dealing with this, but the question was asked in IRC about whether there was a SIP or [unclear] contact only the Tropo sample apps and I guess he is handling that but apparently it is the case.

There is a SIP address.  Now go ahead and quiz that in the IRC in a little bit.

I was going to read all these numbers but the fact is if you are listening to this recording just go to VUC.me and I will probably show all these information and that’s what the site is supposedly for, which is the matter of me getting off my lazy and doing the actual work, so I will try to make sure that all information is in there.  And let me just do a box check here, Dan how much is 57 plus 1.

43.

Well, we have been for about 58 minutes.

Can I speak one last question

Well there is no cut off, I want to make sure that people will understand that we should probably get the questions happening.  Go ahead

Jason, what would happen if one of my scripts take up too long to, if I was retrieving from Yahoo weather and that it was taking 30 seconds. Is there a way to tell the system sort of something that really gets hung up, just fall back on some other recorded message?

What you can do that within your application actually, there are ways to do that programmatically and have a time out within your script that would go ahead and make sure that if that was taking too long such as an HTTP time out or any number of things, we can leave that to you in your application.  We do have some monitors that are running in the background because for example we have had users not intentionally to go out and accidentally write an infinite loop and there is a way for them to go back because they are on a general programming language but we do have ways to monitor those processes, trigger alerts into our own network operation center and then either contact the developer to say, hey what’s up, is this on purpose, if that’s not, we will go ahead and have the ability to kill those individual scripts if it is clearly something that’s just not doing anything.

Thank you.

In your aspect we should mention too is that one of the things we tried to do to help make a platform something that developers can work on very quickly [unclear] we do provide support for free as well, you can just send an email to support@tropo.com and our team will be back to you and will help with that.  We have a live chat that people can interact with as well.

And that is 7/24 and we try to answer any free developer account within two hours.

Actually, it may even be a click to call, that would be really amazing.

There is that too.

But we don’t want to sell you something I think, so…

Okay [unclear] web based widget for interacting with the call?

That’s one of the things we are working on right now.   We actually made another acquisition over there in the UK recently by the name of Clack Point which brings to us a flash cell phone which we are working to integrate as both available to developers as another widget as well as that application you see on the front page, just dial in with your flash cell phone.  So that’s absolutely part of what is in the works.

In production accounts, what do you, do you sort of like paying $50 amount and it sort of ticks away then refreshes on my credit card?

That’s exactly we have gone for, once we are into production it is literally as simple as you will enter your credit card details, you will get a radio button on each of your applications that it should be in development or production, you flip it over to production and it starts decrementing your prepaid balance and then you may have a one time payment, so once the decrement is down, that’s it, you are done, maybe you have done a campaign or something like this, or you had the ability to set re-charge rates at say 10% of the balance, I want to recharge and keep it going.  Yes.

So it is a prepaid similar to a Skype model.

So per applications, in one account I can add three applications where one is a campaign, where one is the money once an allotment is used up, I won’t refresh the card but other applications will refresh the card?

It is actually by account right now.

Oh by account? Ok.

But you can have multiple applications in a single account decrementing that same value but the setting is at an account level but the recharges are not.

So I can sign up multiple accounts.

Yeah, exactly.  You can have an account [unclear] to campaign, put the app in there and put the money in there.

Yes.  I was saying if there are any other questions, we have been going an hour, we can go for three hours if you want, and if you guys have that much time, but as long as there are still questions, I am sure they will….somebody was about to say they have to go?  Any questions?

I am going to have to head out shortly Randy but Jason can stay long but I have..

You will have a party with Jason, I know.

I only came for the after party so.

Which codec [unclear] protocol that you guys are [unclear]?

Yeah, right now we do g 711 and g 729, correct me if I am wrong, on those codecs today.

But g 729 does involve another license, I am not sure if how we work that out with [unclear].  We are working on other codecs as well, stay tuned and you can imagine which ones

Exactly.  But we know

That’s a secret.

[unclear] I have been shown Jason voice chat, I don’t know if you have come across it or heard about it as yet.  On [unclear] that voice chat, the show is Google Wave, but I am attracted by the notion of your [unclear] line and applying it to that because that would be a great test case and from it we could record all the conversations which are helping developers and with the transcription this could become part of a searchable information by which all other developers could access from records of voice conversations, how about that?

That’s very interesting, that’s one thing I was working with Randy on this call as we would love to get this type of conference even recorded and transcribed and put out there, because we love anything that gets indexed by Google out there both for our developers and for marketing and developer outreach and being able to do that in terms of support would be very interesting as well.  Although some time those support calls, we are always nice guys but sometimes the developers can be a little terse.

Well these are case studies and what better case study than a [unclear] developer forum for a voice support capability.

Ah, it is very good idea.

[unclear] put this together.

Absolutely.  I appreciate the input.

What kind of storage capacity are you offering for applications?

Well, the storage capacity, you are writing a script that is going to be consuming our API and that is text based.  What we don’t do today is give you database storage space or recording storage space.  So the for the database side you are going to be talking REST, HTTP most likely to a backend database similar sitting in your network if you need something database driven for our scripting application capability.  If you are using the web API which is the REST JSON interface we have of course you are running in your network in that case.  For recordings and we also do transcriptions, when you ask to record you actually pass us either an FTP URI or an HTTP URI that we can post to or put to and once the recording is complete, we merely post that out to wherever you want to.  We have written example applications out there that show how to push that onto S3 or other places.  So we don’t actually provide MySQL database on the platform, we expect that you be consuming data either through web services or your own backend database in that case.  So one thing to keep in mind, some of the differences with Google app engine is we don’t consider Tropo being an all inclusive general hosting platform, it is a scripting platform to drive our API as opposed to a general web hosting platform.  Does that make sense?

Absolutely.  Thank you.

A Visit to HD Communication Summit near Paris

Google buzz and the privacy concerns surrounding it

General geolocation revelation issues and how location works via wifi

Comparing Amazon, Apple customer service and resolution of issues with Orange. All are very large companies, yet the first two manage to resolve most issues with 24 hours for me (Zeeek) whereas Orange didn’t in six weeks of repeated phone calls.

IRC Links

[5:49pm] ChanServ: [#vuc] Welcome to the VoIP Users Conference (VUC).
[5:50pm] steely_glint: Down side of Buzz – (NSFW F-bombs in the URL) (http://tinyurl.com/ycc3rmc)
[6:28pm] DFrankel: Conference Agenda: http://dev.hdcomms.com/event-schedule
[6:32pm] steely_glint: NSFW Irate blog on buzz (http://tinyurl.com/ycc3rmc)
[6:37pm] RandalSchwartz: … http://www.google.com/profiles/realmerlyn
[6:49pm] kfife: http://www.pddnet.com/news-miniature-six_axis-robot-at-md-and-m-show-021210/
[7:24pm] Zeeek:        http://www.3cx.com/phone-system/downloadlinks.html
[7:24pm] ricko73: http://www.vertical.com/products-televantage.html <–here’s the other one
[8:13pm] Zeeek: dan_lane:  http://www.8774e4voip.com/panasonic_kx_tgp550t04_p/panasonic-kx-tgp550.htm
[8:15pm] maximCH: http://gmailblog.blogspot.com/2010/02/millions-of-buzz-users-and-improvements.html
[9:18pm] Zeeek: you should beconme a fan of VUC page: http://facebook.com/voipusers
[9:55pm] CompleteTech: using http://visualroute.visualware.com/ seems ok
[9:55pm] maximCH: France Telecom http://www.google.com/finance?q=EPA:FTE

LINKS

 Next scheduled session in your time zone
Freenode.net IRC web client, just add your pseudo and you're on #vuc
 VUC Google Group

  VUC linkedin group (business contacts)

Links mentioned in IRC Feb 5, 2010

[6:19pm] NerdUno: Here are some good OpenVPN tutorials: http://pbxinaflash.com/forum/showthread.php?t=4856
[6:21pm] steely_glint: ecrist - http://www.phonefromhere.com/vuc/
[6:31pm] ecrist: Why TCP Over TCP Is A Bad Idea:  http://sites.inka.de/~bigred/devel/tcp-tcp.html
[6:32pm] kfife:        http://www.packtpub.com/openvpn/book
[6:34pm] Zeeek:        http://www.linux4afrika.de/vision.html?L=0
[6:40pm] mfeilner:        http://www.linux-magazine.com/Issues/2009/99/SAFE-CALL
[6:41pm] mfeilner:        http://www.feilner-it.net
[6:42pm] mfeilner:        http://www.openvpn.eu
[6:43pm] mfeilner:        http://www.openvpn.eu/index.php?id=23&L=0

[7:39pm] Skibum: BTW more info on Bria 3 is available at: http://www.counterpath.com/bria.html
[8:00pm] mjgraves:        http://gigaset.com/chagall/provider/general/chagall223_02.bin
[8:21pm] JimCifarelli:        http://www.embeddedarm.com/about/resource.php?item=408

Part 2: Bria Tests and more

From this press release:

CounterPath and Digium(R) Launch Asterisk-Based Multimedia Softphone

Based on open standards, Bria for Asterisk combines Digium’s Asterisk, AsteriskNOW and Asterisk Business Edition telephony platforms with Bria Professional, CounterPath’s enterprise-class, multimedia softphone. Bria for Asterisk gives users advanced telephony features, yet can be quickly set up and configured with any Asterisk deployment.

The VUC is officially testing Bria on Windows and OS X and we hope provide valuable feedback to Counterpath during this launch.

Download audio file (TS-315634.mp3)

Yet another group of geeks discuss the iPad
Download audio file (TS-318318.mp3)

Amy Huson, Director of Marketing at Plantronics, focuses on emerging trends in PC audio for the enterprise. She has over 20 years experience in sales and marketing for audio, video and telecommunications products, software and services. She was the co-founder of an audio software start up acquired by Plantronics and had previously worked in the professional audio division of Harman International, holder of over 50 brands of professional and consumer audio equipment.

Great call!

Download audio file (TS-306310.mp3)

AstLinux is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as OpenSIPS and FreeSWITCH are also available. AstLinux has many features that make it ideal for embedded and commerical Asterisk based solutions.

On the call Friday we will discuss the newly released 0.7.0 version of AstLinux and the number of improvements over the 0.6 series of releases.  0.7.0 will be available with either Asterisk 1.4.29 or 1.6.21 (two separate download images).

In addition to attacks on network availability and authentication, administrators must contend with eavesdropping, audio injection, Caller ID spoofing, VoIP phishing, and other unique exploits. All of these security holes can result in the leakage of information and in unreliable phone calls. Hacking VoIP reviews the many possible VoIP attacks, and discusses the best defenses against them for both enterprise and home VoIP solutions. Author Himanshu Dwivedi introduces popular security assessment tools, describes the inherent vulnerabilities of common hardware and software packages, and provides the first ever VoIP security audit program. The book covers common enterprise VoIP protocols such as SIP and RTP as well as unique protocols like H.323 and IAX.

No Starch Press, publishers of Hacking VoIP, have generously given us two copies to give away to participants of the VUC.

Some of the tools mentioned in the book are on this page.

Download audio file (TS-313655.mp3)

Ward Mundy’s Nexus experience and much, much more in the second hour of VUC from Jan 15, 2010.

Congrats to Maxim and Karl for winning copies of the book Hacking VoIP. I’m looking forward to Dan York’s new book about this subject as well, and hope he’ll join us when it comes out.

Andy Abramson (VoIP Watch), Tim Panton (PhoneFromHere.com), Michael Graves (Graves on SoHo VoIP), David Frankel (ZipDX.com), Ward Mundy (Nerd Vittles), Dave Michels (PinDropSoup.com),  Randal Schwartz , Dave Chan and others discuss the year 2009 and VoIP and the usual digressions.

At the end of session one you can hear the New Years Greetings from listeners and participants.

Session 2:
Download audio file (TS-308412.mp3)

Congrats to Alex Robar, winner of the Polycom IP 335 from e4Strategies

We’ll be live on Jan 1, 2010 and you’ll hear the greetings sent to us from all over the world.

Thanks to everyone who participated from all over the world representing not one, but two species!

Part Zero, pre-conference chat: Download audio file (vuc20091218-Part0.mp3)

IRC Transcript 2009-12-18

Guest Alex Balashov of Evariste Systems leads us down the path to enlightenment and understanding of this complex topic, including

Kamailio, the role of Asterisk, the role of SER and media gateways, etc.

- Relationship of Kamailio to OpenSER project history.
- What is Kamailio/OpenSER?

- SIP proxy
- SIP server (for certain purposes, such as registrar, presence user agent, etc.)
- Common uses of Kamailio.
- Service delivery platform engineering and Asterisk scaling using Kamailio.

- Some discussion of sip-router.org initiative.

Download audio file (Part2gen.mp3)

• 425 Transcoded G.729 calls on a single PCI card.
• Floating licenses for multi-server deployments.
• Support

Howler Technologies has offered to give away 3 packs of 10 g729 licenses in hopes of getting community feedback. Contact them by going to the web site at http://www.HowlerTech.com. If you’re interested in g729, be sure to listen to this informative discussion.

Download audio file (TS-295587.mp3)

IRC Transcript 2009-12-11

Next week: Kamailio and Open SER with Alex Balashov, Evariste Systems

BabyTEL

One of BabyTEL’s features is its single-line/multiple location capability, giving users the ability to route a single number to two or more phones that can be separated by a few feet or by thousands of miles. BabyTEL is in partnership with TalkSwitch, a small niche business service provider of both VoIP and analog small PBX type systems. BabyTEL is also a pioneer in social-networked VoIP having launched the an early integrated telephone service for the Facebook platform called Telephone.

Download audio file (TS-295586.mp3)

IRC TRanscript 2009-12-04

FreeSentral is has three components :

• The Yate telephony server program.
• The Web server that provides the interface.
• The database server that stores all information.

Also starting this week, we hope to introduce a new feature, possibly monthly, with Alex Balashov of Evariste Systems, a consultancy specialising in VoIP service delivery platform engineering, integration, and deployment for enterprises and service providers, including carriers and ITSPs (Internet Telephony Service Providers). Alex has participated in the past and we’ve been discussing a feature that would take VoIP FAQ and toss them around for better understanding.

Download audio file (TS-285273.mp3)

Download audio file (TS-297755.mp3)

IRC Transcript 2009-11-27

Thanks to Digium andfor allocating more channels of Skype for Asterisk to reach the ZipDX conference bridge and to John Todd for making this happen! We had a lot of live callers, including many on Skype.

Unixdawg on FusionPbx
Download audio file (TS-288078.mp3)

This Friday, Allison chatted with regulars Michael Graves and John Covici about procedures and equipment used to provide the quality work with fast turnaround that make her a renowned figure in the voice-over world.

There was also some talk about Skype for Asterisk, which PhoneFromHere.com is using to help make it easier to get into the wideband conference bridge. My own experience on the iPod is that the audio is of excellent quality. I’ve listened on it to several concerts for long durations, it not only sounds good but seems pretty stable, considering that Skype alone is often known for dropped calls.

Also discussed: Broadvoice releases codecs into Open Source

Download audio file (TS-285271.mp3)

Stephen Song of The Shuttleworth Foundation jopined us to discuss The Village Telco Project, an effort to use OSS VoIP hardware and software to supply network & voice services to rural parts of Africa.

The project involves hardware designed by Australia’s David Rowe and manufactured by in China by ATCOM. It runs an embedded Asterisk distribution.

Village Telco is an initiative to assemble/develop the cheapest, easiest to setup, easiest to manage, scalable, Open Source, standards-based, wireless local do-it-yourself telephone company toolkit in the world.  The goal of the project is to render local telephony in developing countries to be so cheap as to be virtually free.  Thanks to advances in Open Source telephony software and the dramatic decrease in the cost of wireless broadband technology, we think this is entirely possible.

The project consists of three principle elements

1. Simplified Billing System for Selling Pay-as-you-go Voice Services

   This element will make a contribution to the excellent A2Billing Open Source project. Work on this project will result in a guided installation wizard for people with limited technical skills and a simple administration and billing interface for business managers.

2. Mesh Potato – A low-cost wireless mesh device you can plug a regular phone into

This is described in the Mesh Potato project page. In a nutshell, the Village Telco needs an affordable device to connect customers to the meshed WiFi network. The Mesh Potato will dramatically reduce the cost of a Village Telco startup

3. Easy-to-use GIS-based Mesh Management Interface

We are partnering with the CSIR-supported Afrimesh project which is developing a fast, easy-to-use, low-overhead management dashboard which enables mesh network operators to create and sustain a resilient communications network with a minimum of fuss.

Part 1: Download audio file (TS-285270.mp3)

Part 2: Download audio file (TS-289673.mp3)

IRC Transcript 20091106

Brian Reynolds visits to explain his part in the demo.

Download audio file (TS-281992.mp3)

irc transcript 2009-10-30.txt

Share on Twitter   

Note: this was a record, 7 hours conference with discussions on many topics, including phones, virtual machines, etc. Will be posting other material soon.

Sound bytes, short sessions in stereo:

Trying to link things together with Google Wave.

Astricon: a few critical comments, a few funny stories…

Download audio file (TS-281991.mp3)

IRC Transcript 2009-10-23

Tim Panton’s ScreenCast of his Astricon 2009 presentation. This used technologies from: phonefromhere.com, wave.google.com, voice.ibook.com, digium.com and skype.com.

Heroic efforts were required to produce this live call and edit it, and it will take patience and strong will to listen to it because of the noise and levels. We post it here because a lot of interesting things were said by participants (in random order) Tim Panton, Darrick Hartmann, Barry Kline, Philippe Lindheimer, Fred Posner, Zoa, David Duffet and whoever else wandered in.

Download audio file (TS-281043.mp3)

All in all, Digium did a great job in organizing the event with compelling content, fun, good food, and a space where people with our shared interests can gather and basically bother Allison Smith for three whole days.

IRC TRanscript 20091016

Download audio file (TS-280550.mp3)

Download audio file (TS-264001.mp3)

Digium AA50 VoIP Appliance to grab at Astricon Phoenix next week!

Digium has just rolled out a new Asterisk speech recognition tool-kit from a partner company called Vestec.  At $99/channel, speech recognition is down to a price point where virtually any application can be speech-enabled.

Also, Fax for Asterisk, Skype for Asterisk, Astricon, mobile phones and more on part two.

Part 1: Download audio file (TS-264000.mp3)

Part 2: Download audio file (TS-275278.mp3)
Transcript IRC-20091002

I’ve had a chance to look through the book, and I’d recommend that anyone who is serious about Asterisk read it as it addresses a lot of areas that have rarely been discussed in a single book.

Considerations about connecting to cellular networks, reflections on knowledge needed for running your own consulting business are just a couple of examples a new material I’ve not seen elsewhere.

Grabbing the list from one of the sites about the book, here are some examples of what’s to be found there:

What you will learn from this book?

* Master advanced dial plans, call routing considerations, and speech technologies.
* Learn all about AGI, AMI, ENUM, and DUNDi.
* Discover how to build an enterprise-scale Asterisk-based solution for mission-critical applications.
* Discover the DAHDI framework, and the way it replaces Zaptel.
* Understand how to apply localization and customization techniques in different territories.
* Learn how to interface Asterisk with analog and digital telephony systems, as well as wireless technologies.
* Appreciate what you need to do as an Asterisk professional when assessing customer needs, Quality of Service, pitching for business, and customer support.
* Understand best practices through example code and detailed commentary.

Download audio file (TS-263999.mp3)

Great call, thanks to David and Nik!

IRC Transcript VUC 2009-09-25 The boys do shine in these quizzes!

Note: From http://voip-info.org “m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option ‘r’, obviously. Use m(class) to specify a class for the music on hold. “

Question: Dial()'m' argument
[7:18pm]Ottone: music on hold for dialtone
[7:18pm] kfife: music on hold class
[7:18pm] ricko73: moh?
[7:18pm] ssokol: music class
[7:18pm] Ottone: it is for "ringing tone"
[7:19pm] J-MN: shouldn't it be the looser gets the book,
    the winner doesn't need it but the looser does...

Port for AMI?
[7:23pm] _3vain: 5038
[7:23pm] ssokol: 5038
[7:23pm] ricko73: 42 <- heh
[7:23pm] Ottone: 5038

Lineup :

• :00 Mobile HD Voice in Moldova
• :02 Fiber to the curb
• :24 Skypecasting
• :28 HDComms
• :37 Nobody cares about VoIP But why they should
• :47 Graves on SL78H
  :50 Aastra Hi-Q Haiku
• :53 OpenPeak
  :54 Snom 870
• :56 Phone Hardware musings

Download audio file (TS-272057.mp3)

GREAT NEWS: Andy is starting a VoIP Watch Podcast.

VUC IRC Transcript Sept 18, 2009

mp3 recording download

Download audio file (TS-267249.mp3)

IRC Transcript 2009-09-11

Official bio: “Aswath Rao has more than 25 years of experience in the telecommunications field, having worked for leading R&D firms. He has worked on ISDN, Frame Relay, BISDN, wireless and satellite communications. For the past 10 years he has been working on VoIP related issues. Long before intelligence at the end became acceptable, he advocated “functional terminals” in ISDN. His proposal for Inter Connect Function has been incorporated in the TIPHON architecture and currently it is known as Session Border Controller. He has developed ways to offer PSTN subscribers many of the features available to VoIP subscribers.”

Take a look at the Slides for today’s discussion

Articles of interest:

On HD Voice, on the supposed Death of VoIP

The En Thinnai project and Innovative Communication Devices

Download audio file (TS-263997.mp3)

Download audio file (TS-267246.mp3)

S675IP and an A58H accessory handset giveaway

Thanks to Gigaset of North America, we had a new Gigaset wideband-capable SIP/DECT phone to give to a participant.

MaximCH was able to identify the singer as Patty Loveless, the song “Waitin’ for the Phone to Ring” and the album. Congrats, Maxim!

We will have another giveaway within a month.

Part 1 Download audio file (TS-256280.mp3)
Part 2 Download audio file (TS-264537.mp3)

Lots of talk about Skype for Asterisk, with Tim Panton who has been on the SfA beta for over a year and who will be speaking at Astricon in October.

IRC Transcript 2009-09-04

Download audio file (TS-261666.mp3)

There is going to be at least one Gigaset giveaway in September so watch this space!

Session Recording:

Download audio file (TS-256279.mp3)

The VUC mailing list is a Google Group

IRC: #voip-users-conference on Freenode.net

irc Transcript 2009-08-21

From the discussion on the Asterisk biz list, a quick look at how these things are financed, how to hook them up and whatever else comes up.

IPKall.com

Sipgate.com (also .co.uk, .de)

Google Voice

Download audio file (TS-256278.mp3)
IRC Transcript 2009-08-14

We referred a lot to the HD Connect Now site.

This week, Doug Mohney joins us. Doug has clocked more than 20 years of experience in the ICT (Information and Communication Technologies) arena between working in real world businesses and writing about them.

His resume includes Editor in Chief, Telecom and Digital Media Group for a B2B emedia company and Editor-in-Chief at VON Magazine (the pulvermedia incarnation, not the Virgo one). Publications he’s written for over the past dozen years include Boardwatch, Mobile Radio Technology/Urgent Communications, The Inquirer, and VON Magazine, covering telecommunications, the Internet, and online video.

His hands-on/real-world experience includes stints at two high-tech start-up companies, one that went public (DIGEX) and one that didn’t (SkyCache/Cidera). Joining DIGEX as employee number 10 in 1993, Doug had a ring-side seat to the trials and tribulations of a fast growing, VC-funded startup doing the boom growth years of the Internet.

Some of Doug’s articles.

Some of the topics:

What is a client, what is a dialer?

- Symbian and iPhone platform differences.
- Voice Quality issues
- SIP Stacks in use (built in, own made)
- Hand out / Hand in
- Presence
- Chat
- Video support

and a lot more. I did end up buying and using iPico SIP client on the iPod Touch G2 and works best of the three apps I have tried so far. See the post just below this one for details.

Part 1: Download audio file (TS-249092.mp3)

Dave Nelsen, founder and former CEO of Talkshoe to joined us for a long fireside chat. Dave has an extensive background in telecommunications and it was great to get his take on today’s situation. He also spoke extensively about Talkshoe’s technologies used in PSTN, VoIP and on the web.

Download audio file (TS-251758.mp3)

Is Skype about to become history? Dean Collins began a discussion about news he got from an Australian paper about the Ebay/Skype rumors. It seems when they bought Skype, Ebay The Devil didn’t purchase the core technology?

The Asterisk users list has an interesting thread where religious convictions opinions about POTS vs SIP (SIP vs Analog lines) are being flung right and left as usual. Since it’s a lazy summer, at least speaking for myself on vacation, we can kick this around among us as well.

Also, Maxim in Zurich mentioned an interesting site on Twitter: Gizmo Voice where youcan see the tricks needed to get a free Google Voice phone number and connect it to your Gizmo number to make and receive calls to/from USA numbers. You can also use the Gizmo Call site to make calls with your Gizmo account using a Flash client on that web page.

Download audio file (TS-244093.mp3)
Download link 2009-07-24

Voxeo Labs’ Jason Goecke, Jay Phillips and Dan York  join us to talk about the news:

Ok, the word is out, Voxeo Labs is born! Voxeo, provider of Unified Communications and Self-Service platforms, announced that the Tropo.com cloud telephony service source code will be made available to developers for free under open-source licenses. The new Tropo source release demonstrates that cloud computing vendors can subscribe fully to open-source ideals, and avoid the proprietary lock-in typically found in cloud services. Tropo is the first of several new open-source projects from Voxeo Labs, the innovation and open-source focused organization announced by Voxeo yesterday at OSCON.

From TheNextWeb.com : Don’t accept vendor lock ins!

More links and Tropo info.

IRC Transcript 2009 07 24

Download audio file (TS-244090.mp3)

We spoke to Ruben Olsen who will be a guest on August 7th to talk about mobile VoIP clients. Freenum.org was an extended discussion. Lots of other stuff. Some interesting talk with Leif Madsen about general dialplan tricks (recipes). First installment: How to read a file in the 1.6 dialplan.

Thanks to Tony for arranging the A580IP giveaway contest. Congrats to regular Karl F for winning the phone!

Download audio file (TS-235829.mp3)

Michael Graves and I have both talked about how we think these phones are a revolution for the SoHo market. How will they be supported? Who will set them up, provision them, get them working with different  pbx and SIP providers? Michael’s most recent blog post talks about the Gigaset DE380IPR, the inexpensive wideband capable desk phone.

Our guest today is Anthony Stankus, Gigaset Communications product Manager for North America. The Gigaset IP DECT products which include: A580IP system (base and handset), S675IP system (base and handset), A58H accessory handset, S67H accessory handset, SL78H high-end accessory handset.

Tony will be with us on the ZipDX wideband conference bridge to answer these questions and any others you can collect.

More on these phones:

Latest mgraves.org on the new Gigaset phones

Graves On SOHO VoIP » Newsworthy: Siemens Gigaset S675IP

Alan Lord’s The Open Sourcerer.com

My own modest consumer review

Siemens Phones – voip-info.org

Steven Perich » Blog Archive » Siemens Gigaset IP phones

Siemens Gigaset S685IP VoIP phone (Personal Computer World)

DECT phones arrive

IRC Transcript 2009-07-10

Thanks to Dave Michels, who met him at Glue Con, Twilio CEO Jeff Lawson and CTO Evan Cooke join us to discuss their “cloud telephony” offering. You can prepare by getting a free account and checking out what can be done with the platform. Note that speech to text (transcription) is a part of it.

“Twilio provides an in-cloud API for voice communications that leverages existing web development skills, resources and infrastructure.”

Twilio are giving away a free netbook book each week to developers that build cool voice applications: http://www.twilio.com/contest/netbook/

Jeff Lawson will be speaking at OSCon July 20-24th later this month about open source, telephony, and Twilio.

Spend an hour and a half with Matt Florell talking about VICIDIAL, the Open Source Call Center software, and plenty of other subjects including CentOS, Ubuntu, resource issues, hardware for robust asterisk systems.

About VICIDIAL:

VICIDIAL is an enterprise class, open source, call center suite in use by many large call centers around the world.

VICIDIAL has a full featured predictive dialer. It can also function as an ACD for inbound calls, or closer calls coming from VICIDIAL outbound fronters. It is capable of inbound, outbound, and blended call handling. VICIDIAL even allows you to have agents logged in from remote locations.

IRC Transcript 2009-06-26

Part 1: (Talkshoe) Download audio file (TS-224823.mp3)

Part 2 (ZipDX recording served from CloudFront): Download audio file (ZipDX20090619.mp3)

We’ve tried to get people to talk about this before, but we didn’t get a deep explanation. Here’s a chance to ask Nir, who did a compelling presentation at AMOOCON on Asterisk on EC2, to clear up any of those nagging questions you might have.

Nir’s AMOOCON presentation, Dynamic Asterisk Scalability with Amazon EC2 and videos are available on the AMOOCON site. Nir’s company is Greenfield Tech.

Nir is also the author of Asterisk Gateway Interface 1.4 and 1.6 Programming

Jerry Shuman of Perssonas (@theagent) joined the call with some great input, too.

IRC Transcript 2009-06-19

Follow on Twitter

@voipusers @e4voip @mjgraves @viperdudeuk @steely_glint @teamforrest @fredposner @asteriskbot

Transcription

Nir Simionovich: I have to admit that the first person that got me heavily into EC2 is actually here and it’s John Todd. I’ve been dealing over the past two years a lot with Asterisk and Virtualization mostly VMware, and one day on the list there was this question about Asterisk and VMware if it’s doable, what is the performance, and I actually replied to it, and John came up with the idea, “Well, there’s been some talk about EC2 and Cloud computing,” and I said, “Well, why not? Let’s give it a try and see what we can do.” As time progress and what’s really progressively fast00 was the time of the Israeli elections and I was working with a company in Israel that was doing outbound dialing to get people to come into to vote. So, we ended up building the entire infrastructure of EC2 because in order to generate as many calls as we needed to, there was simply no infrastructure big enough here in Israel to hold like 32 servers. So, we ended up opening multiple EC2 accounts and doing everything from EC2 and it worked really well and that actually sprung everything into like real rapid motion and it got me heavier and heavier into EC2.

Zeeek: I think the first question in most people’s mind before they know anything about EC2 and I don’t know how many people know much, I don’t know a lot. I know about S3. I’ve used it. We use S3 and CloudFront of services extensively as somebody just posted the Amoocon talks, that nearest talk is number 27 and if you’re looking in IRC, you can see that, the link, otherwise, just find the talk number 27 on Amoocon.de.

Anyway, Nir, the first perception of EC2, you have to wonder because for years, Asterisk has been this full thing. You’ve got a Linux box and you download it and you install it. It’s fairly easy to install if you have any experience in Linux at all, and then the thing is working and you got it on your machine and you play with it, and you know, it’s free and blah blah blah, but when you’re doing an EC2 Instance of it, I guess one of the main reasons to do this is because it’s inexpensive but it’s extensible, right? During the elections if there were thousands of thousands of calls being made, you’re not going to be able to do that on your old 386 that you threw together, right?

Nir Simionovich: No. That would have been impossible. The thing was that we actually needed a way to expand really fast. Now, we have an infrastructure here in Israel of servers. We have like six different tools beyond servers and because we were doing also media and outbound dialing and we’re handling a lot of capacity here in Israel, we got limited by the number of concurrent calls we’re able to per machine. So, sure we’re able to push it up to about 240 concurrent calls, 260 per server. So that wasn’t big deal but we needed more and the idea of using the EC2 just sprung up saying, “OK, let’s see what we can get out of that specific system.” So, we ended up going over there initiating what’s called the medium AMI. That’s AMI based on, I believe it’s a dual core and it has 8 gigs of RAM and 160 gigabyte of disk, and we created our own AMI image, sprung those up and each one was able to get about 120 concurrent channels on it with media and we’re really happy because we could now expand as much as we want.

As you said, it’s pretty easy to get Asterisk up and running on your own machine. It’s pretty easy also to get Asterisk up and running on Amazon EC2. The main problem after you get it up and running is to make sure that the work that you’ve done actually stays there because the one thing that, for me at least, is pretty annoying about Amazon EC2 is that once you bring up an instance inside EC2, if you shut it down, it will simply go away. It will get deleted. So you have to maintain your work somewhere else and going about on expanding your system with EC2 requires a fairly new skill set that most Asterisk developers currently don’t yet have, but it’s like completely mashing up the entire voice world that Asterisk is pretty much dominant in the web world where most, let’s say, IVR developers are not that accustomed to working with.

Fred: Hi, this is Fred. I was wondering on 260 things, you said that you’re limited to about 260 calls per server and that seems a little low for a dedicated hardware back and I was wondering what kind of setup you were using or what the limiting factor was.

Nir Simionovich: Well, actually the limiting factor was not Asterisk itself. It was actually the number of file descriptors we had opened on the box and the network that we had, and the boxes themselves were just dual cores. These were old IBM x335 servers and we were getting about 240, 260 concurrent calls, concurrent channels with media which were interacting with the database and external sources and the application itself was pretty heavy. So, we ended up running into a load average on the box itself around four, 4 ½ and we didn’t want to breach that one. We could have gone higher. That is for sure. That shouldn’t be a problem. Yeah, we could have gone higher to about 360, 400 but we didn’t want to. We were afraid of actually harming the quality of the call.

Man 3: Yeah. I’m curious if you’re using Amazon Elastic Block Store for persistent drive space for any of this architecture and if you’re noticing any sort of timing issues with it?

Nir Simionovich: Well, I have to admit that we hadn’t used it. We didn’t see any reason to do so but the minute I’ll do a test of that, I’ll be sure to publish those.

Man 3: Thanks.

Zeeek:  I can kind of see what EC2 is about but as far as Asterisk goes, first of all, I assume that someone has—you have the possibility now just finding instances that are already built and kind of throwing them up there? In other words, rather than starting from the beginning, someone’s already done this work, correct?

Nir Simionovich: That is correct. There was a lot of work done by Eric Chamberlain from Voxilla. The main issue that EC2 suffered from, the main kernel issue that each of the ES, let’s say, the standard AMI images that EC2 used was a different clock being compiled into the kernel and that caused issues with things about, let’s say, conference calls. It will just come up wrong. What Eric did is create a new image that is based off a 1,000 hertz clock which is pre-set. I think its 1,000 kilohertz, sorry. The thing was that—what Eric created was something really optimized for Asterisk to be working off Amazon EC2 and that made a lot of things in a lot of the work. It took a lot of the edge of working with Asterisk and the EC2 because it came in a pre-packaged AMI. You can just take it, you can modify it, whatever you want, and you can get your systems up and running really, really fast.

I know of a few companies that actually managed, like these virtual call centers. What they do is actually, they initiate small instances of EC2 according to the customers they have and what they do, they just initiate the instances as they go along. That means that if somebody right now is working and they need like four hours to do an outbound campaign or 10 hours to do an inbound campaign, then they’ll just bring out a proper EC2 image saying, “OK, work off of that,” and they will just rented you a call center per hour and it’s a completely new way of thinking about how to do call centers and how to visualize PBX systems and host PBX systems. So it brings a lot of possibilities back into the mix.

Zeeek: Jim is asking you, Nir, in IRC about how you distribute the load among multiple EC2 instances. It’s a good question. I’m trying to picture this stuff and it’s pretty foreign to me.

Nir Simionovich: Well, basically what we did, we used carriers that we’re able to distribute the load according to what we needed. We just gave them all the IP numbers, all the public IP numbers of our EC2 Instances and they actually distributed the loads for us, but we could have also easily taken, let’s say, an open sear, put it in front of the system and use that to distribute the load to each of the instances.

Zeeek: Other question I had was with regard to configuring. So, I don’t even understand how you access this. This looks to you like you’re going to login to it and it just looks like any box, is that it? You go with SSH into it?”

Nir Simionovich:  …basically, Amazon EC2 is based on Xen.  So, essentially what you’re getting is, you’re getting your own Xen domain.  Basically, imagine that, if you’re familiar with Xen, then imagine that EC2 is like this enormous Xen dom zero that operates, and whatever host, whatever AMI you issue, or you initiate, you go about and you actually create your own dom inside that, let’s say that big cloud.  Now what you get is your own virtual machine.  So, you get a machine that you can SSH into it, and work with it, and do whatever you want.  It behaves exactly like a normal machine.  Sure, you have no control, well, you have some level of control about where it is located in the world, in terms of geographically saying, “Okay, I want to be in this specific data center or in that specific data center”.  But, again, that level of control doesn’t give you a lot of ability to say, “Okay, if I initiate two, or four, or five different AMI instances”, that they will be located in the same area.  And that is the main problem with using EC2, because the normal, let’s say, coupling of servers that we are very much accustomed to when building Asterisk environments doesn’t exist there.  There is no ability to say, “Okay, if I initiate database, then the database is sitting right next to the application, and there’s no latency between those two.  So, if you do that, that’s kind of a hassle.  So, you need to go about, and say, “Okay, we need to decouple our application.  We need to decouple our information storage”, and say, “Okay, we’re going to move everything outward, into a completely new location, and have Asterisk work off of that”.  And that requires new skills.

Zeeek:  No question about that.

Nir Simionovich:  By the way, somebody on the IRC just said that Open Series is now called Open SIPS, that’s not entirely correct.  Open Series is now referred to as Camie-Camiellio[sp?], and Open SIPS is a fork of Open [unintelligible], so these are two different things.

Dean Collins:  Randy, it’s Dean Collins.  I just wanted to jump in.  So, are you going to set up a consultancy practice offering this type of technology, and if so, what’s the URL for your company, so people know how to get in touch with you?

Nir Simionovich:  Well, okay.  The easiest way to get to my website is [omitted, since it was an incorrect address].

Zeeek:  We’ll have to remember to post that in the…

Nir Simionovich:  Yeah, well actually, it’s…

Zeeek:  …BBC site.  Oops.  Ooh, bad URL.  Bad.

Nir Simionovich:  [laughs]  Yeah.  Or, if you’re able, well, because most of the audience here are English speakers, it’ll be a lot easier if you go into www.greenfieldtech.net, then you’ll get to the same place, it seems.

Zeeek:  How did John Todd get involved in EC2, by the way?

John Todd:  Because it seemed to be fairly obvious, that in order to do something really big that scaled, that EC2 is a good platform for it.  There have been, previously there have been some people doing work with EC2 doing experiments, really.  But kind of a formalized test with actual results that’s been kind of lacking.  We still don’t have really good, quantifiable tests with EC2, saying we can handle this many media channels on a particular instance, before we start to see pattern problems.  And, well, really, the whole point of my talking with Neer[sp?], actually, and a couple of other people, was to see if we could get some quantifiable results on Amazon’s network.  Because, of course, Voice Over IP is very subject to things like packet loss and jitter.  I wanted to see if Amazon’s network was going to hold up to Asterisk, or RTP, in general, operating over their fairly large infrastructure, and all, I guess, non quantifiable reports seemed to indicate that it does work just fine.  But I still am having…

John Todd:  I don’t have any actual numbers saying that, you know, for these three days, we tested RTP streams, and we saw this packet loss to this provider.  My goal is to get Amazon to offer, and this still may happen, we had some brief conversations with the folks at Amazon, but I’d love to see them offer a service where service providers, meaning PSTN termination providers, could interconnect with Amazon’s EC2 infrastructure at no cost, and those voice packets would be transmitted at no cost to the instance users.  In other words, don’t pay for the bits in and out of Amazon’s network if you’re using this particular service provider, who is bearing the burden of network interconnect.  That would make this much more appealing for VoIP services, and application providers to outsource their activity to Amazon, and I think that that’s really just waiting for kind of a critical mass of EC2 infrastructure stuff to happen…

Zeeek:  And of course, you were channeling Don Quixote, as always, John.

John Todd:  Yeah, to some degree.  But actually, Amazon is pretty clueful, if they see this as a way to increase the number of hours people use the machines, then I don’t think that, you know, their network costs, I think, are pretty close to break even, unless you’re doing something like video.  But I think anything you’ve got to do to encourage people to get onto their system, they would be interested in.  One of the big services that is based on EC2 and Asterisk is Toolio[sp?].  All of their infrastructure right now, from what I’ve understood them to say, is that it is running on EC2, scalably, with Asterisk as the back end.  And, as a matter of fact, they’re going to be giving a talk about that at AstriCon this fall, in October.

Zeeek:  There was a question about, and a very good question from Jim, about bandwidth cost calculations.  Now, you did give us some idea, Neer[sp?], during your presentation about the costs.  Why don’t you run over that a little?  Whatever you can, or want to reveal about the cost of the campaign, some of the numbers.

Nir Simionovich:  Well, the calculation is based on…well, okay.  Let’s start from square one.  Amazon EC2 economics, the way they calculate your usage is a little different.  That means that they calculate your usage according to your instances.  For example John, I’ll just punch it up for a second.  For example, they would go about and say, “Okay, for every gigabyte that you consume, up to a certain level, you will pay something like, let’s say a cent”.  Now, if you know that specific calculation, and you’re able able to calculate what is your projected traffic, and you can calculate you’re projected traffic according to the codecs you use, and how many concurring calls you have, and how many sessions you…and how much time you will be operating the system, you’re pretty much easy to say, “Okay, I can go about and say, okay, I’ll be utilizing, let’s say 800 gigabytes over the next two weeks of traffic”.  So, you’re able to say, “Okay, 800 gigabytes of traffic, multiplied by one cent, that’s eight dollars worth of traffic”.  So, that’s how the math actually works.  Now, to say that there is a calculator that enables you to calculate that exactly, it’s a little hard to say.  I don’t think that one exists, but the math, in itself, isn’t that complex to do.  In general, just use any type of bandwidth calculator that you’ll be using for any VoIP application, and just calculate your, let’s say, megabit per second ratio, and once you have that, just multiply that by the number of seconds you have over a course of your operation, and you’ll have your number.

Zeeek:  It’s interesting, Dave EG[sp?], from EGW, we have a customer who uses EC2 for their telephony apps, multiple Asterisk instances, front ended by Open SIPS, you have some logic which brings up and down additional Asterisk instances for peak load times, which is basically, Neer[sp?], what you did during that election campaign.  Interesting.

Nir Simionovich:  That’s exactly the same.  Same idea.

Zeeek:  Yeah.  No question, this is great for things that you pointed out, and one of those was, hey, you’ve got this project that lasts, what was that, three days, is that right?  Something like that?

Nir Simionovich:  It was three days worth of dial out, and it was operating about ten to twelve hours every day.  So, we got out about 36 hours, 38 hours worth of dial out.

Zeeek:  It would be incredibly hard to put together a hardware platform, and stupid [laughs], to put together a hardware platform for three days, unless, you know, you worked in a hardware store, or something, to do this project of over a couple of days.  Even if it was a few weeks, whereas this is one of the things where…

Nir Simionovich:  That’s exactly the case.

Zeeek:  The whole EC2 thing shines.

Nir: This is exactly what it pays. Awhile back, John and I did some mass going about and saying, trying to understand if there is a validity to actually going about and running a host PBX on top of Amazon EC2, and we realized that if you operate for 24 hours, a full system is operating 24 hours, then there is no real usage for Amazon EC2 as is. If you’re a business which is doing a hosted environment, then Amazon EC2 is the best you’ll find, really. It gives you that immediate quantifiable capacity that is really, really fast,but if you’re on business, let’s say, if you’re in an office and needed a PBX system, then Amazon EC2 isn’t really the thing for you because you can get an equivalent server with inside dedicated hosting facility today for like $59 a month and that one already has 16 gigabytes of traffic with it, maintenance, managed hosting and so on and so on. So, some services make sense working off of EC2, some don’t. If you’re going to say, “OK, we want to move into the Cloud,” that specific move has to be calculated and has to be tested, let’s say, economic wise.

Zeeek: The agent has posted—now, I’ve seen calculators for S3 before and this is it, right? Well, I’m going to read this because I don’t have time to make a short URL, Calculator.S3.AmazonEWS.com/calc5.html.

Nir: That will give some insights as to how much money you’ll need to pay.

Zeeek: I think it’s worth trying a project, maybe messing with it to see. So once you’ve got your instance ready to go, I mean, if the elections are held next weekend, are you ready to bring that same instance up? What’s involved in bringing that back to life? Can it be done?

Nir: Yeah, it’ll take about five minutes.

Zeeek: So where is this thing stored? I mean, it’s in your account and you just click, you just check a box and click submit and you can bring it back up or is it something locally stored somewhere with you or what?

Nir: It’s exactly how you described. What you do is, once you build your own box, you can actually create your own AMI image that contains all your software. That’s exactly what we did. Once we had our own AMI image, the private image, we can then store inside S3 and what will happen is, the minute we may want it, we’ll just go into our AWS account and we’ll just say, “OK, we need now five instances of this specific AMI running,” and we’ll just click that and punch it up immediately. No big deal.

Zeeek: Interesting. The agent is talking about the ElasticFox extension and start playing with EC2. I have absolutely no notion of what that is but hopefully, either you do or he’ll call in and tell us about it.

Nir: Yeah. ElasticFox is a Firefox extension that enables you to play with Amazon EC2 instead of working with the web interface. It’s only for Firefox, that’s one. I think a combination of both ElasticFox and the web interface as is because the new web interfaces they’ve created which is currently and still in beta is, in my view, just a bit easier to use than the ElasticFox, just a bit.

Zeeek: OK, and as I mentioned, IRC, I use a Firefox plug-in sometimes for S3 or for CloudFront. So, I assume it’s a similar concept. You just kind of dumps it down.

Nir: Yeah, similar concept. Yeah.

Zeeek: It’s kind of amazing to me that—and this goes back to what we were saying with John Todd that there’s no good reason to think that Amazon’s network would allow for real time. The CloudFront thing, the idea there is that you can be serving video and stuff like that and that it will push it out to the edge servers that will be the most efficient. On the other hand, I’m not sure how that translates to VoIP and I’m surprised to learn that they’re able to do the real time. Who knew, you know. That’s the way I feel about it. So it’s kind of surprising that they have the capacity—you don’t have any problem with call quality apparently, right?

Nir: No. Well, let’s say this, as long as you’re working with carriers which were interconnected with, let’s say…

[Music playing]

Zeeek: Yeah. Let me try to find the source of that.

Nir: OK.

Zeeek: That may have been Digium. Hello, Digium. Thank you. They may not. Sorry.

Nir: That was a good one.

Zeeek: Yeah. Well, that’s the Asterisk built-in…

Nir: We can talk over the music.

Zeeek: Frankly, I think that was somebody in Huntsville or else I just happened to click it by accident. Sorry. That happens from time to time.

Nir: Yeah.

Zeeek: Where were we?

Nir: In any case—where were we? Yeah, in any case, we were working with three or four different providers in the U.S. which are considered somewhere around, let’s say the tier 2. Then, usually, these carriers are interconnected on the backbone with links that are very, very efficient and reliable over to Amazon EC2 as long as you’re hosting your instances in the U.S. So, the end result that were, let’s say one of the carriers that we were using was called TSG Global and we were getting about 24 millisecond roundtrip from Amazon EC2 to TSG Global. Now, you’ll run voice over IP on that and will have perfect quality always.

Zeeek: I’m trying of course the agent to call in, trying to figure out because it would be more interesting to have a discussion here than reading the stuff in IRC.

Nir: Yeah, absolutely.

Zeeek: OK. My arguments have won over. He’s going to call. He or she, you never know, right?

Nir: Well, he brought up a very interesting point of how to bundle your own AMI and this is actually what I was talking about, creating your own AMI image. It’s not that simple. I have to be honest. It took me about four or five hours to actually understand how to do it right and it never works from the first try. It’s really annoying. What happens is that you need to take your running machine, your running AMI and you’re creating a raw06:25 image of the hard drive, of the virtual hard drive and then you upload that into S3. That’s complicated to do but if done right it will work on the second or third try. It never works on the first try, never.

Zeeek: OK and we have a bleep from Southwest British Columbia. Can you hear us? Could be the agent, you never know.

Nir: Actually, I’d love to hear about what he just wrote on Openfire XMPP and Red5 Flash Server for video conferencing. That is really interesting because I’ve been trying to play around with Red5 but I have to admit that Flash isn’t really my thing. So, I’d love to hear about that.

Zeeek: OK, the agent is coming out of the wild of Southern California. Who is in the Southwest British Columbia? Say hello. Come on, don’t be shy. We’re waiting for the agent to call in. We have time.

Man 2: What do you do for your logs then here, just logs if you’re just bringing up these instances and using for so many hours and shutting them down? Are you uploading those at some point, back up to S3 or do you just not care about them?

Nir: Well, the application as is doesn’t really carry that much logging into the actual server itself but I have to admit that I never really needed to go about and offload the logs anywhere because we don’t really are about those. CDR’s are kept completely separated from the system but you could offload those into a different machine or you can, before shutting down the system, go back and offload those out of EC2. It shouldn’t be that much of a problem. Or if you really want to, you can use the—it’s not S3—what was that again—it’s called…

Zeeek: CloudFront?

Nir: No, not CloudFront. They’ve got servers where you can actually map a storage device to your AMI and then you can share…”

Nir Simionovich: This is exactly what it pays. Awhile back, John and I did some mass going about and saying, trying to understand if there is a validity to actually going about and running a host PBX on top of Amazon EC2, and we realized that if you operate for 24 hours, a full system is operating 24 hours, then there is no real usage for Amazon EC2 as is. If you’re a business which is doing a hosted environment, then Amazon EC2 is the best you’ll find, really. It gives you that immediate quantifiable capacity that is really, really fast,but if you’re on business, let’s say, if you’re in an office and needed a PBX system, then Amazon EC2 isn’t really the thing for you because you can get an equivalent server with inside dedicated hosting facility today for like $59 a month and that one already has 16 gigabytes of traffic with it, maintenance, managed hosting and so on and so on. So, some services make sense working off of EC2, some don’t. If you’re going to say, “OK, we want to move into the Cloud,” that specific move has to be calculated and has to be tested, let’s say, economic wise. Moderator: The agent has posted—now, I’ve seen calculators for S3 before and this is it, right? Well, I’m going to read this because I don’t have time to make a short URL, Calculator.S3.AmazonEWS.com/calc5.html. Nir Simionovich: That will give some insights as to how much money you’ll need to pay. Moderator: I think it’s worth trying a project, maybe messing with it to see. So once you’ve got your instance ready to go, I mean, if the elections are held next weekend, are you ready to bring that same instance up? What’s involved in bringing that back to life? Can it be done? Nir Simionovich: Yeah, it’ll take about five minutes. Moderator: So where is this thing stored? I mean, it’s in your account and you just click, you just check a box and click submit and you can bring it back up or is it something locally stored somewhere with you or what? Nir Simionovich: It’s exactly how you described. What you do is, once you build your own box, you can actually create your own AMI image that contains all your software. That’s exactly what we did. Once we had our own AMI image, the private image, we can then store inside S3 and what will happen is, the minute we may want it, we’ll just go into our AWS account and we’ll just say, “OK, we need now five instances of this specific AMI running,” and we’ll just click that and punch it up immediately. No big deal. Moderator: Interesting. The agent is talking about the ElasticFox extension and start playing with EC2. I have absolutely no notion of what that is but hopefully, either you do or he’ll call in and tell us about it. Nir Simionovich: Yeah. ElasticFox is a Firefox extension that enables you to play with Amazon EC2 instead of working with the web interface. It’s only for Firefox, that’s one. I think a combination of both ElasticFox and the web interface as is because the new web interfaces they’ve created which is currently and still in beta is, in my view, just a bit easier to use than the ElasticFox, just a bit. Moderator: OK, and as I mentioned, IRC, I use a Firefox plug-in sometimes for S3 or for CloudFront. So, I assume it’s a similar concept. You just kind of dumps it down. Nir Simionovich: Yeah, similar concept. Yeah. Moderator: It’s kind of amazing to me that—and this goes back to what we were saying with John Todd that there’s no good reason to think that Amazon’s network would allow for real time. The CloudFront thing, the idea there is that you can be serving video and stuff like that and that it will push it out to the edge servers that will be the most efficient. On the other hand, I’m not sure how that translates to VoIP and I’m surprised to learn that they’re able to do the real time. Who knew, you know. That’s the way I feel about it. So it’s kind of surprising that they have the capacity—you don’t have any problem with call quality apparently, right? Nir Simionovich: No. Well, let’s say this, as long as you’re working with carriers which were interconnected with, let’s say… [Music playing] Moderator: Yeah. Let me try to find the source of that. Nir Simionovich: OK. Moderator: That may have been DGM. Hello, DGM. Thank you. They may not. Sorry. Nir Simionovich: That was a good one. Moderator: Yeah. Well, that’s the Asterisk built-in… Nir Simionovich: We can talk over the music. Moderator: Frankly, I think that was somebody in Huntsville or else I just happened to click it by accident. Sorry. That happens from time to time. Nir Simionovich: Yeah. Moderator: Where were we? Nir Simionovich: In any case—where were we? Yeah, in any case, we were working with three or four different providers in the U.S. which are considered somewhere around, let’s say the tier 2. Then, usually, these carriers are interconnected on the backbone with links that are very, very efficient and reliable over to Amazon EC2 as long as you’re hosting your instances in the U.S. So, the end result that were, let’s say one of the carriers that we were using was called TSG Global and we were getting about 24 millisecond roundtrip from Amazon EC2 to TSG Global. Now, you’ll run voice over IP on that and will have perfect quality always. Moderator: I’m trying of course the agent to call in, trying to figure out because it would be more interesting to have a discussion here than reading the stuff in IRC. Nir Simionovich: Yeah, absolutely. Moderator: OK. My arguments have won over. He’s going to call. He or she, you never know, right? Nir Simionovich: Well, he brought up a very interesting point of how to bundle your own AMI and this is actually what I was talking about, creating your own AMI image. It’s not that simple. I have to be honest. It took me about four or five hours to actually understand how to do it right and it never works from the first try. It’s really annoying. What happens is that you need to take your running machine, your running AMI and you’re creating a raw06:25 image of the hard drive, of the virtual hard drive and then you upload that into S3. That’s complicated to do but if done right it will work on the second or third try. It never works on the first try, never. Moderator: OK and we have a bleep from Southwest British Columbia. Can you hear us? Could be the agent, you never know. Nir Simionovich: Actually, I’d love to hear about what he just wrote on Openfire XMPP and Red5 Flash Server for video conferencing. That is really interesting because I’ve been trying to play around with Red5 but I have to admit that Flash isn’t really my thing. So, I’d love to hear about that. Moderator: OK, the agent is coming out of the wild of Southern California. Who is in the Southwest British Columbia? Say hello. Come on, don’t be shy. We’re waiting for the agent to call in. We have time. Man 2: I’m here. Moderator: Yeah. Man 2: What do you do for your logs then here, just logs if you’re just bringing up these instances and using for so many hours and shutting them down? Are you uploading those at some point, back up to S3 or do you just not care about them? Nir Simionovich: Well, the application as is doesn’t really carry that much logging into the actual server itself but I have to admit that I never really needed to go about and offload the logs anywhere because we don’t really are about those. CDR’s are kept completely separated from the system but you could offload those into a different machine or you can, before shutting down the system, go back and offload those out of EC2. It shouldn’t be that much of a problem. Or if you really want to, you can use the—it’s not S3—what was that again—it’s called… Moderator: CloudFront? Nir Simionovich: No, not CloudFront. They’ve got servers where you can actually map a storage device to your AMI and then you can share…

Open Fire XMPP and the Red Five. Let’s get into that. You guys… Sure. …go ahead. I mean, I’d love to hear about that one. Yeah, so. Well, let me, let me, kind of, lay out the landscape a little bit for why I would even go down this path. I don’t know if you’ve heard of a specific podcast or video podcast that’s done and has been done for a couple of years, called Gilmor Gang It’s run by a guy named Steve Gilmore who’s a well know technology columnist here in, or up in Silicon Valley. Anyways, I’m basically an executive producer of that show and for the longest period of time we’ve been trying to put together, you know, a remote TV studio on the fly. And the biggest thing with this scenario without doing any hardware was really the concept of having, having remote individuals around the world that could dial in, in whatever manner they had available to them. If they had the ability to use a video teleconference capability either through a flash or through a Skype connection or if they just had a basic call-in functionality we needed to incorporate all those things because we wanted to see, we want to use a remote video teleconference capability as an actual remote camera so that we can bring them into the show. We can switch in and out of the different guests because generally Gilmor Gang handles 4-6 different guests on a given show. There’s a lot of banter. So, really, this has been an ongoing investigation of mine for probably over a year. And I’ve been using EC2’s primarily for, at this point, for Personis itself. That is completely hosted on EC2 all the way, all the way through. S3, EC2 SQS. I mean, I probably use almost every service floating around inside of Amazon at this point. And, I also had played a number of months ago, using, putting Asterisk up and running it there, playing around with it to see how well that would work. Which, it works extremely well. I mean, you’re able to take advantage of Amazon’s significant infrastructure, their tremendous pipe, that they have going throughout their data centers and their multiple peering relationships that they have for Internet access. So, I was very pleased with that. So we’re moving on to the next go-round. Okay. Hold on. Sorry about that. It’s okay. It’s hard to predict. But once I see the wave front, it’s over. Go ahead. Anyways. I thought, I thought the angels were coming for me or something. So. The a, so anyway, we’ve gone to this capability in an attempt to basically create this real-time audio-video capability that can be brought up or down on the fly. And that’s, kind of, the real story. Because, I mean, as he was mentioning regarding the phone, the service that they were running where it’s, they only need it for a short period of time. That’s the extreme value of Amazon. I mean, that’s where you really, really, really can take advantage of the system. Because you only need to bring up an instance for as long as you need it and then you can bring the instance down. You know, you can use it in such a, in that manner for a [R?] escape function, so that if you have, you know you’re going to have a burst of traffic coming in. You know, depending on how you set it up, you can pre-stage servers sitting there waiting for the burst or you can even do, you can get a little more sophisticated and create an auto-scale mechanism that, as traffic builds, it automatically instantiates new instances to handle the traffic burst and shuts them down accordingly as the traffic falls off. So, I mean, if you looking at, just to give you a price understanding, from an Amazon perspective. Just on a small instance. If you’re running an instance. The smallest instance that Amazon has. It costs you approximately, and this is without, you know, significant bandwidth going in or out. It’s only $70, it’s about 70 bucks a month. Now, the real significance, though, is that if you don’t need it up all the time, take it down. And that’s, kind of, what we’re doing with this whole remote video communication capability, in that, we can pull-up or pull-down anybody actually. We’re setting it up so that we can have remote. If somebody else wants to do a show that we want to add to our network, we’re going to have this all pre-configured so that they’ll have, they’ll have video-conferencing, they’ll have, and that’s the whole Red Five to XMPP integration. So we have our own chat functionality through XMPP. Don’t need IRC. We also have all the video capability that is, and SIP capabilities that are inherent inside of XMPP itself with Open Fire. And then we have the complete telecommunications capability that we’re embedding with Asterisk. And then, of course, the Red Five stuff, which is really handling the, you know, the multiple videos simultaneously function. That’s why the Red Five exists. So that I can have a room, literally, of ten, you know, video, teleconference, or video pieces coming in from all the participants. So those are the types of things that we’re currently working, or I’m currently on, that hopefully, we’ll see as, basically, a package that we’re going to load out into the world, that people can run. And that’s the other part of it. You can take advantage of the really big thing about the AMI’s, the other big part that I take a lot of, that I take advantage of all the time, is just that there’s a lot of people out there packaging these AMI’s. They, if they put them out in the public space, you can just go and click on one of them and say, “hey, what is this one, oh, look, there’s a Red Five server already configured by somebody. I’m going to go ahead and launch that AMI.” You can customize that AMI to your heart’s content and then ultimately save that to your, save that yourself as a private version of that instance. There’s a vast amount of power sitting within the Amazon infrastructure to take advantage of.

Nir: Jerry, one small question in regards to Red5, and this is something I have been toying around with, and maybe you’ll have an answer, because I didn’t get much work into it. Are you working with the, by any chance, with the Fontventa extensions for 3G, 3.5G cellular video transmission? Jerry: No I’m not. It’s an interesting… I’ve seen the extensions, I haven’t played with them at all. That would be kind of a next evolution for us. The other thing is that right now I’m kind of using, it’s kind of a limited version of Red5, in the sense that I’m actually running, and I didn’t even know this could exist until I found the ability through some board. There is actually, you can take Openfire and you can actually run Red5. Somebody’s written an extension that actually runs Red5 inside of Openfire. So, you don’t need to set up multiple servers. It’s basically an instance that’s prepackaged and you can basically bring it up and you’ve got full blown Jabber and you’ve got full blown Open5 sitting there, I mean Red5 sitting there, ready for you to use. The extensions that could take advantage of 3G, 3.5G, and 4, I mean those are all things that would be very, very cool to be able to put in. Because as you can see with things KiteTV and QIK, and a bunch of the other ones, you know there is no doubt that kind of capability from a video teleconference capability is going to be big, and big in the very near term.
Nir: I have to admit that I didn’t have much time to play around with… Well l did play around with the Fontventa extensions, and I’ve done some work with that, in trying to integrate video, IP Video cams and Asterisk, and have, I’d say, some fair success with that. The thing is a lot of people now are talking about it. They are actually integrating a flash environment back into your cellular phone over 3G instead of writing your complete IBPR and doing that through Red5 seems like a good idea. But I will have to admit, that I hadn’t been playing with that lately. Again it’s very reassuring for me saying – OK there’s actually somebody out there really doing it. I mean, really progressing in that direction and we’re doing that on top of EZ2 is really exciting. Jerry: Yeah, and we think it’s really powerful, too. Again, the concept is that once this is a kind of an operational tapestry that we’ve kind of, or are orchestrating, that we really plan on bundling it as a kind of a preconfigured service that people in our… You know the concept, as we can all tell, media is changing for the worst, for the better, however you want to look at it doesn’t really matter to me, but bottom line is this entire users interface content and community that’s floating around out there and taking over, and watching mainstream media die. I mean, the idea that we can have this full blown, on the fly network capability brought up at will any where in the world and have multiple correspondence with basically multiple instances of both video and voice, there is an exorbitant amount of power there, especially when you start looking at what’s happening with the real time, in the real-time space, with things like Twitter and others. So, I think this whole new real-time space is actually going to usher in with Twitter and others, the ability to usher in this whole new paradigm for real-time, instantaneous, dynamic virtual, in essence radio stations, television stations, that can pop up on the fly and shut down on the fly at will. And it’s all going to be run through, basically, a cloud. So, that’s kind of what I’m focusing in on.
Nir: Well I totally agree that there is a shift. Where we can all… well you know people who are already in it and are dealing with the shift, they are all feeling it pretty strongly. I remember when they started talking about cloud computing here in Israel that actually, the thing is that I was in this, there’s the Grid Community here in Israel, they were doing this event and I was looking at it saying – Well it’s pretty interesting but it doesn’t do much for real-time. Then I got back home and started to talk again with John Todd and saying – Well why not do it? It’s pretty interesting that there are real usage of real-time applications that can be done on the cloud and that’s really interesting to me, and really exciting. Not only from a commercial point of view or the economical point of view, but also from a technical point of view, because we have to rethink every thing. Where we’re really used to using like this enormous servers and saying – OK I’ve got 8 cores, I’ve got 16 cores, I’ve got 32 gigabytes of ram, and suddenly we don’t have that and it brings up a lot of the old paradigms that I used to work on back in the days of lets say Commodore 64 and saying – OK we need to be something really, really optimal, and the optimization is really interesting. It’s bringing back really old paradigms in terms of optimizations and really new paradigms in terms of economics.

Jerry: Yeah, yeah. I agree with you. I think what’s been happening, I mean it’s kind of how software leads hardware in some ways. You know, over the course of the past decade or more, the extreme focus on, or at least in the places I’ve played, a very laser guided focus on abstraction and decoupling. I mean that’s been a big part of a lot of systems that I’ve been a part of in the past 10 years. I think now it’s interesting to see cloud computing and the way it’s basically being pushed out from a hardware standpoint and how all the aspects of abstraction and decoupling are instrumental in being able to do these systems. Because you really do need a fairly significant abstractional error because you don’t know where the hardware exists. You don’t know how much memory you’re necessarily playing with. So, this whole decoupling is pretty paramount but it’s also been a trend that’s been going on for at least a decade or more.

Nir: Yeah, it’s been around here but not many people were using it, you know. I started my, let’s say, I started using the main frame computer somewhere around the age 18 or 19, when I was in the Army, and decoupling applications and the world of the main frame just sounds logical. It’s there, there’s nothing else. But when you are used to working on PC’s and servers, decoupling applications looked really weird. When you come to the company saying – OK you need to decouple this and they go – What? What are you talking about? What I don’t have access to the data base? That doesn’t sound right. The paradigm shift is going in a few directions and I’ve seen some companies that they, company like Giga Spaces, it’s an Israeli company, and they build like this enormous middle layer between your application and the Amazon EC2 cloud, which gives you the, let’s say, the imaginary effect of working inside a coupled system while you are completely decoupled. Which is really interesting. Jerry: Yeah, yeah.
Nir: There is a lot of work being done right now. I think we are heading, in terms of technology, we’re heading for the next, I believe, is like the next three years is going to be really interesting in terms of cloud computing. Jerry: Yeah. I think the biggest thing for me, I’m somewhat of serial entrepreneur, and have been that way for a couple of decades now, but I think the thing that really excited me about the advent of things like EZ2 or Rackspace’s mossomo, or any of the other ones that are floating around there. There’s Slicehost which actually is now owned by Rackspace. It’s just the cost at which somebody could bootstrap up something, an idea, a concept. To be able to sit there and create a new business in this new paradigm and do it at a cost that is unheard of. Having done a few startups in my lifetime, you ultimately throw out an inurnment amount of money upfront just to handle, from a capital standpoint, the hardware that you are going to run on, especially if you’re a service offering. So I mean, that’s one of the things here, where you can get away with some, a couple of weeks worth of coding, you know, a $70 fee and you’ve got a service up and running in an hour or more, or a little bit more. But it’s nothing. It’s nothing to do and it’s cost effective, and if it doesn’t pan out, you basically shut down the AMI and you walk away from it.

Today we talked about the EXPERIMENTAL link using Skype for Asterisk (SfA for short). The more we use SfA, the more the Skype people will see that we the Asterisk community are serious about it.

The whole thing began as a phone call made by Tim P. to my Skype number. I almost never have Skype on but the number is forwarded via Gizmo5 to my phone, so when I heard it ring and saw the Skype ID, I realized the call was going through a bunch of servers and trancodings. The quality of that call was striking: total silence when no one was speaking (the equivalent of very black on an LCD display) and very good quality audio with a comfortably small latency. The call lasted at least 30 minutes, too and remaind of good quality.

So, we had at least 6 people called in via Skype from the USA, Israel, Barbados and the UK. The quality varied with the number of channels in use and time, but it was pretty good by and large.In fact, the quality when there was no packets dropped sounded good, even though it was g711.

IRC Text Transcript 2009-06-12

Download audio file (TS-224821.mp3)

Transcript IRC 2009-06-05

Open a free Sipgate account and check out what they offer so your questions can be answered and your opinions aired!

Above is the Sipgate user panel. Sipgate offers 200 free minutes to test their system and a free USA phone number. Free calling between Sipgate extensions, of course but you can put several phones on a single account. I did just that and then called myself (from Europe to Europe) and found the lag to be as expected but what a great solution for newbies and people who want to experiment.

Sipgate does voicemail as expected but also, Google Voice-like ringing of multiple extensions. Try it out if you dare.

Part 1:

Download audio file (TS-224820.mp3)

Part 2:

Download audio file (TS-229501.mp3)

Jim Van Meggelen is one of the author’s of O’Reilly’s Asterisk: The Future of Telephony, and an old school PBX guy. Jim is probably a bit of a masochist, which would explain why he got into the telecom business in the first place, and why he now loves Asterisk. Jim is pretty friendly, kinda like a puppy that gets your shoes dirty. His enthusiasm is infectious, but also a little bit frightening if you stand too close. Jim is a partner in Core Telecom Innovations, a recognized leader in the Asterisk Telephony Revolution, and iConverged Inc. He lives in Toronto with his wife and three kids, and loves writing, photography, speaking, improv, choral singing, and old shoes.

Jim will talk about building an embedded-like system using x86 hardware (Intel Atom Mini-ITX) and a stock Linux distro for well under $1000, including FX cards with hardware echo cancellation — this is a great way to build a system that’s almost like an embedded system, but with much less software complexity or incompatibility.

IRC TRanscript 2009-05-29

Part 2: Yate and g722 Download audio file (TS-227343.mp3)

YATE with Diana Cionoiu, who always shoots from the hip. Yate is an open source software developed with reliability, scalability and flexibility in mind for tomorrow’s telephony networks. I guarantee an interesting call this week, but then we always have interesting calls, right? Otherwise, why would they last up to four hours like today’s call?Here is a link to Diana’s presentation at Asterisk Tag 2008. She was also at AMOOCON 2009, here’s the page for her presentation. (Almost all the presentations are now available on the AMOOCON site. Nice work, Stefan!)

Acting upon the suggestion of our group, here is the IRC transcript of the call which again lasted over 4 hours.

IRC Transcript 2009-05-22

Part 2: Yate and g722

Askozia PBX is an embedded PBX solution which eases system upgrades, backups and provisioning. Currently running under FreeBSD, they are in the process of porting it to linux, which is also worthy of discussion. (File under trials and tribulations!)

I met Michael at AMOOCON and invited him to join us after he told me a funny story: someone had sent him to a VUC session recording that had a laugh track sounding after a mention of Askozia (or was it FreeBSD?). Anyway, I do recall that the laughs were for our friend Rich, not Askozia. We would never make fun of an Asterisk-based project in that way. Except if Rich is behind it . Just kidding, Rich!

Watch the AMOOCON video of Michael’s presentation is available here thanks to Stefan Wintermeyer’s quick work.

I apologize for mis-pronouncing Michael’s name throughout the discussion, it’s pronounced “eeee-demah”-, not “Eyedemah”.

Part 2:

Download audio file (TS-224860.mp3)

Part 1 MP3 DOWNLOAD LINK

Part 2 MP3 DOWNLOAD LINK

IRC Transcript 2009-05-15

A great time was had by all, John Todd and I hash over some of the highlights. Part two: Diana and Paul from YATE.

Download audio file (TS-222070.mp3)

MP3 DOWNLOAD PART 1 MP3 OWNLOAD PART 2

IRC TRanscript 2009-05-08

Although I managed to herd a great group of people into the speakers’ lounge and record a wide-ranging discussion, the first part was lost due to a software glitch on the recorder. It seems that the low battery level shut things down and ditched the file. The part posted here is only the second hour (heh) but there are some interesting things to hear.

Topics covered include codecs, video, Skype for Asterisk, teleconference robots, “robot Mark”, presence, stereo codecs, and lots more. One side (Zoa, Diana Cionoiu etc) rags on the iPhone, the other side says “get used to it”.

MP3 DOWNLOAD LINK

I had the pleasure of meeting one of the more vibrant and outgoing characters at the conference and recorded this interview about his latest PHP AGI book. I only wish I could have shared the stories, jokes and general mirth happening in the traditional German restaurant when Tim Panton, Jason Goeke, Nir and I shared a meal.

Nir’s presentation at AMOOCON was about running Asterisk under EC2 and I hope to get him to talk about this some day as he has an actual large-scale case he deployed in this way rather than just a theoretical one.

Nir’s book, Asterisk Gateway Interface 1.4 and 1.6 Programming is a great introduction to serious programming using the AGI and PHP. The book takes you through some good solid advice for truly professional programmers on how to do stable programming in the Asterisk AGI environment. This is the book you need if you are trying to do anything serious with Asterisk in PHP.

We talk about home automation and random asterisk and telephony news, including the possible final demise of Nufone. On the home automation front, it seems every product uses proprietary systems so little progress has been made with anything standard. We were lucky enough to rope in an expert on the subject so listen to what Tony has to say on the recording of Part 1. X10, ZigBee  and many other systems are discussed by people with hands-on experience.

http://delicious.com/voipusersconference for all the links discussed. A lot of names were mentioned so check the links.

IRC Transcript 2009-05-01

Daniel Berninger – Washington, DC based independent technolgy analyst. Expert in technical and regulatory aspects of Internet enabled disruptive communications. Active in VoIP since 1995. Daniel worked on the original assessment of VoIP at Bell Laboratories and led early gateway deployments at Verizon , HP, and NASA after joining VocalTec Communications. He won the 1999 VON Pioneer Award as co-founder of the VON Coalition and worked on the founding of ITXC , Vonage , and Free World Dialup . Daniel gets quoted frequently on regulatory, antitrust, and VoIP matters.

Dan’s last visit on September 14, 2007 is here.

Here are a few articles about wideband:

Wideband Audio Codecs Gaining Steam,

Wideband Audio Boosts VoIP Quality

G.722, WideBand Audio

Download: Part 1 Part 2
IRC Transcript 2009-04-24

Randal Schwartz (FLOSS Weekly) chatted with us about paid support for Open Source software.

Steve Sokol joined us to talk about what this means to the community and to small, medium and large busines installations.

With Digium’s support subscriptions for open source Asterisk, enterprise developers and systems administrators can call on the expertise of the company that created Asterisk. Digium’s technical support team offers professional assistance with installation, configuration, and systems operation issues.

Digium’s blog has a post about this, and that sends you to the product page.

A lot more happened including discussion of Digium’s new free FAX channel that replaces drop in solutions such as SpanDSP and the eternal questions about SfA (Skype for Asterisk).

Note the name siptosis, unlike the similar word “halitosis” is pro-nounced “SIP 2 SIS”. These calls are educational!

IRC Transcript 2009-04-17

IRC Transcript 2009-04-10

If you’re into telephony whether for business or pleasure, you need to know more about the service Google finally rolled out. I’ve been testing it for the past few weeks (along with Gizmo5) and I hope others will bring their input, too. The current version features groups, with separate vmail announcement messages for each group (and even for each person). It can screen calls, send you an email transcript, record calls, allow you to listen to incoming vmails, send an SMS alert, etc. Your incoming calls on your lifetime number can be routed to any number of phones, cells, Gizmo accounts and even… GTalk.

Part II: Download audio file (TS-212912.mp3)

Try the web click 2 call. Enter a name and your phone number:

The name will be read to me if I’m listening in. Or, try to leave a message and let’s see what the transcription looks like!

We also talked about a call I was on where a deaf person spoke with several people by using sign language to an interpreter through his webcam and the interpreter skpoke the words to us on Talkshoe. Amazing! He was using viable.net which is apparently a free service.

Check out the voipusersconference Delicious bookmarks page for more related to each session.

IRC Transcript 2009-04-03

I’ve been testing all the ways I can find to connect between phones and Skype. I happened to be testing OpenSky when I saw some posts from Michael Robertson on the asterisk-users mailing list. As I exchanged a few emails with Michael, I realized he had been a founder of Mp3.com and well as Gizmo5.

He was generous enough with his time to spend a full hour (on a cellphone!) answering questions and sharing thoughts about the various SIP to Skype solutions, including Digium’s Skype for Asterisk, Gizmo’s OpenSky and Skype’s SIP for Skype which Michael calls vaporware.

Post conference chat, aka “Cognac”

The post conference has discussions about the expectation that all services should be free with David Frankel of ZipDX.com Randal Schwartz from FLOSS Weekly and the regular gang. Due to some problems with Talkshoe, various audio sources are used so you can hear what different phones sound like.

Download audio file (TS-210205.mp3)

Download audio file (TS-198844.mp3)

IRC Transcript 2009-03-27

SIP hacking has escalated and there was a wave of it in recent times. John Todd published a simple common sense set of rules that are worth reading. One of the main risk areas is using user names and secrets that are too easy to guess (such as extension 200, username=2000,password=wakeme).

More on SIP for Skype etc. That looks like it will be an ongiung discussion. See the session with Michael Robertson above.

We briefly reviewed a new book called Asterisk Gateway Interface 1.4 and 1.6 Programming. John Todd and /me have both been looking at it and we both think it looks good. Since the discussion, I’ve read more of it and I will talk more about it in a future session.

Amazon link (beer money!)
Asterisk Gateway Interface 1.4 and 1.6 Programming