Panel discussion on the topic of cloud computing, with several experienced players in this field.  We aim to start with some basics about cloud computing, then explore its various uses, including telecom applications.

Guests with cloud-related experience include:

• Eric Chamberlain, Founder of RF.com, presenter to Astricon 2009 on running Asterisk in the Amazon Cloud
• Jason Goecke, Creator of the Adhearsion Ruby framework for Asterisk, representing the Tropo service offered by Voxeo.
• Greg Weidenhammer, VUC regular, works for HP on massive scale systems
• Tim Higgins, Publisher of Small Net Builder and the new site Small Cloud Builder
• Aretta Communications*, Provider of virtualized Asterisk services

*participation not yet confirmed

Readwriteweb has a list of interesting podcasts about cloudcomputing.

Asterisk SCF is a framework that will allow developers to move beyond voice and also create open source video and text that can be used for anything from embedded applications to enterprise and carrier solutions, according to Digium. It is delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, and will support pretty much all real-time IP communications including video, wideband audio, chat and desktop sharing.Asterisk SCF seeks to meet four requirements for businesses looking to deploy open source communications solutions:

• Performance
• Scalability
• Fault tolerance
• Extensibility

Live from Astricon (well it was when we recorded it), Kevin Fleming, director of software technologies at Digium joins us at talk about the big Digium announcement.  This discussion was made possible  by Tim Panton and Michael Graves. Kevin, Tim and Joshua Colp were in a room, surrounded by John Todd’s tie-dyed shirts.

Thanks to those who requested this on our mailing list! We have liftoff! (VoIP Security)

VoIP Abuse Project . J. Oquendo

“Arkeos is a program primarily used on Unix based servers running the Asterisk Open Source PBX. The application mimics a valid extension in which an attacker, after bruteforcing an account, attempts to place calls. What the attacker doesn’t know is that their calls go nowhere. The sole purpose of allowing them to perceive they have an account is to track them. Where they come from, what accounts they use, what VoIP account scanners they use, what numbers they try to dial. Because of the flexibility of Asterisk and Unix as a whole, the application does a lot more than stated however, I will not disclose too much otherwise any attackers reading this might catch on. Then I won’t have anyone to play with.”

From Jason Goecke’s blog post – read the whole post here on the Tropo blog. Jason made the announcement to the world about this new project on the VoIP Users Conference!

“We have decided to take it a step further and created a Tropo Scripting application that turns Tropo into a giant Asterisk application platform in the cloud.   You can now run just about any Asterisk AGI application on Tropo.

Tropo AGItate was started on the Nerd Bird (good to have in-flight WiFi) from San Jose to Austin, on my way to LoneStar Ruby Conference. Jim Freeze – the organizer of LSRC – had recently been to AdhearsionConf in San Francisco; I wanted to be able to show something extra special during my talk there. On that one flight, I was able to get the basics working and show Tropo emitting AGI during my talk, just like that. (For those non-Asterisk folks out there, AGI is an API that lets external applications connect in to Asterisk and fully control it).”

Key points:

• Tropo can now speak AGI over TCP to any FastAGI server, including Adhearsion, PHPAGI, Asterisk-Java, etc.
• A long list of Asterisk commands are supported, and of course all the Tropo ones.
• While the script is written in Ruby, no Ruby knowledge is necessary to use it. Just point your Tropo application to the Ruby script on Github, upload a configuration file via FTP or WebDAV to your Tropo account, and you are ready.
• Full support of Tropo Speech-Synthesis (TTS) and Speech-Recognition (ASR), but also for Asterisk sound files. Yes, tt-monkeys works!
• Fail over to a SIP URI, in case your FastAGI server does not respond. This could be to another Tropo application, another Asterisk box, or anything that supports SIP.
• You get all of the Tropo channels over AGI, including SMS, Instant Messaging and Twitter, all using the same application.
• Support for custom SIP Headers in and out of Tropo.

“High Definition Voice” on phone conversations is something you may not notice until you have experienced either very bad or very good quality. Andy Abramson of VoIP Watch will be joining our discussion of HD rollouts by [a carrier whose name begins with O] and the ramifications and challenges as he expressed them in his recent article.

Allison Smith (@VoiceGal) pinged me about the collection she recorded for Joey Lindstrom for use with Asterisk pbx. The full set, which Mr. Lindstrom has generously decided to place in the public domain is available for dowload free here:
http://vuc.li/FunnyAllison.

Joey wrote to say this was his way to give something back to the community, and he’s asking Digium to put these prompts in the Asterisk distribution should they wish to do so.

2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set.

Video testing – Next week we’ll be doing a real video conference, watch the site for more info

ClueCon - No one wished to comment. Whatever.

Both SIPVicious and Adhearsion have announcements to make. Sandro joins us to explain “svcrash”, a new addition to the tool kit to stop attacks when they happen:

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

Friday June 18th we will be extending our recent trend in discussing matters of VoIP security.  This call will feature Ken Kuenzel from Acme Packet. Acme Packet are the leading provider of “Session Border Controllers,” aka SBCs.

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.

John Todd gives a preview of Astricon 2010, we chat about AMOOCON in Rostock next week and about various VoIP trials and tribulations. We have a special VUC discount code for you. Enter code AC10VUC for a significant discount.

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.

Among others, Ward Mundy (Nerd Vittles, our guest next week) and many of the VUC regulars join in this violent argument civil discussion about who is responsible for filtering, where it should take place and the how and why of their ideas on the subject.

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.

Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP phones.

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.

We talk about several topics in part 2 including Skype, USB connections for VoIP, new devices like the Gigaset One, a cellphone gateway. Lots of talk about NAT with Asterisk and SIP. The usual chatter from friends around the table in our weekly “club”.

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.

The recording is from part of the ZipDX recording of April 16, 2010.

The Media5-fone is a SIP Client (softphone) that enables users to make and receive VoIP calls. VoIP calls are calls established over a Wi-Fi connection using the IP technology of the Media5-fone.

Tim Panton and David Burgess describe the project to deploy the first GSM phone system to the small island nation of Niue.

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.

[audio:http://recordings.talkshoe.com/TC-22622/TS-338121.mp3|titles=Sam Reports on Voipathon]

Mihaela from Token d.o.o. joins us for a chat about Sam Reports, custom software from her family company located in Croatia. She and her husband have been running an Asterisk integration shop since 2007 and writing custom software to ease and enhance working with Asterisk.

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.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-317306.mp3|titles=VoIP Security]

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/

VoIP online community and VoIP how-to

Voipathon will be talking about online community, dealing with censorship, countries that intentionally block telephony to prevent communications and much more. Using voip, connect with your online community and accomplish great things (or just socialize, we do both!)

Join us for Voipathon Live!

Honolulu: Thursday 25th at 9 PM
San Francisco: Midnight Thursday night to Midnight Friday
New York: 3:00 AM Friday
France,Germany: 8 AM Friday
Mumbai: 12:30 PM Friday
Beijing, Singapore: 3 PM Friday
Sydney: 6 PM

VoIPathon will be held for 24 hours, beginning on March 26th at 3 AM Eastern Time. Subjects discussed: online communities doing good things via the Internet. Online community helps you find like-minded people to learn from, share info and get things done.