Category: Open Source
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What Devices And Software Support The Opus Audio Codec? Here Is A List
Continue Reading: What Devices And Software Support The Opus Audio Codec? Here Is A ListWhat devices support the Opus audio codec? What softphones? hardphones? call servers? Obviously given that Opus is the “mandatory to implement” audio codec for WebRTC, it will be in many web browsers… but what other I was asked this question by a colleague recently and when I couldn’t easily find a list on the Opus codec web site, I turned to the VUC community inside of Google+ and posted there. The great folks there naturally were a huge help, and quickly came up with this list: UPDATE: No sooner had I hit “Publish” then I discovered that Wikipedia has a list of devices and software supporting the Opus codec. As that list is much longer than this one below, I’d encourage you to look at that list.
- Web browsers supporting WebRTC:
- Google Chrome
- Mozilla Firefox
- Softphones:
- Blink
- Counterpath Bria
- Jitsi
- Linphone
- Mumble
- (Maybe Skype? They talked about it.)
- “Hard” IP phones:
- Mobile applications:
- Acrobits (iOS)
- Counterpath Bria (iOS)
- csipsimple (Android)
- Switches / Call Servers / IP-PBXs:
What other devices or software supports the Opus codec? (Or what other lists are out there listing devices supporting the Opus codec?) Please do let me know either by comments…
- Web browsers supporting WebRTC:
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Today’s VUC Call – Setting Up A Cellular Network In The Desert For Burning Man
Continue Reading: Today’s VUC Call – Setting Up A Cellular Network In The Desert For Burning ManToday’s VoIP Users Conference (VUC) call at 12:00 noon US Eastern should be quite an interesting one. Tim Panton from Voxeo Labs and Tropo will be joining the call to talk about his experience setting up a mobile network in the middle of the desert for this year’s Burning Man event.Tim recently described the experience in a guest post at TechCrunch: “What We Learned Running A Mobile Network At Burning Man” and on the VUC call will talk more about what he did. The FAQ from the Papa Legba camp at Burning Man makes for quite an interesting read. I’m looking forward to hearing more from Tim… and the call is open for anyone to join in.
You can join the live call via SIP, Skype or the regular old PSTN. There is also an IRC backchannel that gets heavy usage during the call. It will be recorded so you can always listen later.
If you found this post interesting or useful, please consider either:
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Today’s VUC Call All About The “FreeSWITCH Cookbook” – Noon US Eastern
Continue Reading: Today’s VUC Call All About The “FreeSWITCH Cookbook” – Noon US EasternToday at noon US Eastern on the VoIP Users Conference (VUC) Call for Friday, April 27th, the group will discuss the brand new “FreeSWITCH Cookbook“[1] published by PACKT Publishing. The four authors of the book, who are also leaders of the FreeSWITCH project, will apparently be joining the call.
While Asterisk generally gets most of the “open source VoIP” buzz, the folks at the FreeSWITCH project have been working away on their own solution. As they will say, FreeSWITCH performs a different role than Asterisk and is used in different contexts.
FreeSWITCH has become quite a powerful platform and I’m looking forward to learning more about what is going on with the project right now.
You can join the live call via SIP, Skype or the regular old PSTN. There is also an IRC backchannel that gets heavy usage during the call. It will be recorded so you can always listen later.
As noted on the VUC page for today’s call, the show will also be simulcast in video using Google+ video and YouTube. If you are interested in joining the video side of the call, please follow the instructions on…
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Video: How to Communicate at Burning Man using OpenBTS and Tropo
Continue Reading: Video: How to Communicate at Burning Man using OpenBTS and TropoHeading to Burning Man this coming week? Would you like to use your mobile phone to connect up with others on the playa in Black Rock City?If so, check out this video from Chris Pirillo about the work being done by a team of folks to supply local cell phone coverage… the vans with satellite and cell hookups are already enroute… it uses software from OpenBTS and Tropo.com to let burners leave each other voice messages, exchange SMS messages and more. Here’s the video:
And here are some blog posts that provide more information:
- The Long and Winding Road to Burning Man
- Tropo + OpenBTS + Burning man = Awesome
- Voice Board and Group SMS for Burning Man
- Papa Legba FAQ (about the deployment at Burning Man and what you need to do to participate)
I’m not personally going to be at Burning Man, but this does sound very cool!
If you found this post interesting or useful, please consider either:
- The Long and Winding Road to Burning Man
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Adhearsion open source telephony framework has new source code repository
Continue Reading: Adhearsion open source telephony framework has new source code repositoryI’ve long been a fan of the work that Jay Phillips did to create the Adhearsion open source telephony framework and so I was delighted to read today of news of its future. To give some context, Jay first created Adhearsion a number of years back because he was frustrated with how hard it was to create dial plans with the open source Asterisk PBX. So Jay went off and created a framework where a programmer could use the Ruby programming language to very simply create voice applications.Jay went on to team up with Jason Goecke to further develop Adhearsion and then last year Jay and Jason joined Voxeo (my employer) to create Voxeo Labs out in San Francisco. While Jay has since moved on, Jason continues to move Adhearsion forward and announced today that Adhearsion has a new home on Github … and hinted at much greater plans in store. I’m looking forward to seeing what all they may be…
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Voxilla Tutorial – Running Asterisk in a EC2 Cloud
Continue Reading: Voxilla Tutorial – Running Asterisk in a EC2 CloudLong-time readers will know that I have been intrigued for a long time with what we now call “cloud computing” (and have written about it and spoken about it) and also continue to find the world of open source telephony interesting.So naturally when I’m pointed to a step-by-step tutorial about running Asterisk in Amazon’s EC2 cloud, I’m interested. 🙂 It’s a nicely done tutorial and I look forward to seeing what people will do with it. (Unlike Mark Headd, who pointed to the tutorial in a tweet, I won’t be trying it out this weekend, but I will be doing so at some point soon.)
Cool stuff…
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Technorati Tags: amazon, ec2, asterisk, opensource, cloud, cloudcomputing
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Has Asterisk NOT “crossed the chasm” for developers? (Key links to read for open source)
Continue Reading: Has Asterisk NOT “crossed the chasm” for developers? (Key links to read for open source)Jay Phillips is frustrated. He passionately wants to see open source telephony enjoy success all around the world. Yet right now, when people think “open source telephony”, they almost always think of Asterisk… and Jay sees too many challenges for developers embracing Asterisk. Jay, the creator of the Adhearsion telephony framework for Ruby, has spoken about this at recent conferences and pulled together his thoughts in a lengthy post earlier this week entitled “What We’re Not Admitting about Asterisk“.Jay argues that Asterisk has not crossed the proverbial chasm for developers and outlines some of the issues he sees.
What is perhaps most interesting about Jay’s post is the equally lengthy response by Asterisk creator Mark Spencer. Mark responds to Jay’s various points and in doing so provides some good insight into his views on Asterisk’s connections to developers, APIs, etc., as well as the differences between the markets that Digium, the company, goes after versus the “market” of Asterisk, the raw telephony platform.
Both Jay’s article and Mark’s response are definitely worth reading. I’m friends now with both of them and they both bring immense passion and energy to the world of open source telephony. Ultimately they…
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Heading out to ClueCon 2008, Telephony Developer Conference, this week..
Continue Reading: Heading out to ClueCon 2008, Telephony Developer Conference, this week..This afternoon I’ll be heading to the airport to fly out to Chicago to be part of ClueCon this week. Haven’t heard of ClueCon before? Here’s the quick summary:ClueCon – is an annual 3-Day Telephony User and Developer Conference bringing together the entire spectrum of Telephony from TDM circuits to VoIP and everything in between. The presentations and discussions will cover several open source telephony applications such as Asterisk/Callweaver, Kamailio (formerly OpenSER), Bayonne, YATE and FreeSWITCH.
Billed as the “Telephony Developer Conference” it primarily focuses on the whole world of open source telephony.
I’ll be there as part of two panels. First, tomorrow I’ll be joining fellow VoIP bloggers Andy Abramson and Thomas Howe on a “VoIP Roundtable” to talk about current industry themes and trends. Then on Thursday I’ll be part of a “VoIP Security Roundtable” talking about… gee… can you guess?
It should be a fun event… I’m looking forward to catching up with Andy, Thomas, Moshe Yudkowsky, Jon Todd and several others. There are also some folks on the schedule with whom I have corresponded but never physically me, so that will be nice as well. If any of you reading this will also be…
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Nortel’s fascinating move into open source telephony… but NOT with Asterisk
Continue Reading: Nortel’s fascinating move into open source telephony… but NOT with AsteriskNortel and “open source telephony”? Huh?That was admittedly my thought when I received the list of who was going to be on the panel I moderated last week at VoiceCon on open source telephony. The other two panelists were obvious choices: Bill Miller was from Digium (makers of Asterisk) and Raza was from 3Com who have recently announced that they would be reselling a version of Digium’s Asterisk Business Edition. Both Bill and Raza made sense to me. But Tony Pereira of Nortel? Nortel does not leap out at me as a company working with open source telephony – what in the world are they doing with it, I wondered?
It turns out that the answer is… “quite a bit!”
As Tony Pereira outlined in our panel as well as in conversations afterwards, Nortel is in the process of launching their “Software Communications Server 500” (SCS 500) targeted at small businesses and built using open source telephony software!
Interestingly, though, it does NOT use Asterisk.
Instead Nortel is using the “other” major player in open source telephony, the “sipXecs” product from SIPfoundry.org. (Previously called “sipX” but renamed “sipXecs” about a year ago.) I’ve not written all that much…
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My presentations at VoiceCon this week…
Continue Reading: My presentations at VoiceCon this week…I’m down in Orlando this week for VoiceCon Orlando and will be part of three sessions. Tomorrow, I’m moderating a panel at 8am on VoIP security and on Thursday I’m moderating a panel on open source telephony. On Wednesday, I’ll be part of a keynote panel with Irwin Lazar on “Social networking and enterprise communication”, which should be quite fun. I’ll include below the full descriptions of the various sessions. If you are attending VoiceCon and want to connect, please do contact me.Session Title: Top VOIP Security Threats
Date: 3/18/2008
Time: 8:00 AM
Room: Osceola B
Session Description: There’s been a lot of concern about voice over IP security, but have there been many actual exploits? This session will inform you about the state of VOIP security. You’ll learn about generalized IP attacks that have affected IP telephony systems deployed on IP networks, and you’ll also find out what VOIP-specific attacks have actually been observed “in the wild”–and what to expect in the future.
KEY QUESTIONS: * What are the most serious voice-oriented attacks that are actually being carried out? What potential attacks haven’t occurred yet but probably will before long? * How do you protect your VOIP systems against…