Asterisk, the global leader in open source IP PBX and telephony!
beelinebill August 19th, 2008
When you’ve been in the networking and telecom industry for any long period of time, you run into many people over and over again and even get to know them quite well as “industry colleagues”. The author of the blog, The Hyperconnected Enterprise, Tony Rybcynski from Nortel, is one of those who has shared panels and hospitality suites with me over the years. Check out his latest assignment on his blog, which is to rationalize why Nortel, with a huge hole in their software product line acquired a tiny open source company called Pingtel which had already been acquired last year by Bluesocket. It is clear Nortel was not controlling the development priorities within Bluesocket and felt it was easier to just “take them off Bluesocket’s hands” this past week. Since it is an open source project called sipXecs they already had the code, so why bother? We find this intriguing.
It seems like yesterday at VoiceCon in Orlando at the CMP hospitality suite when I asked Tony about his new “strategy du jour” (which is really Tony’s evangelist role at Nortel) because at this VoiceCon event he was touting Nortel’s Microsoft partnership and how it was changing the world for Nortel. This is Tony’s new evangelical effort, open source and targeting Asterisk. What I find humorous is Tony has some but little knowledge of Asterisk and he is listening to the rhetoric of a competing open source project who on their web site just has it mostly wrong.
Tony is clearly attempting to evangelize his own product without performing the due diligence to substantiate his position when he boldly claims that Asterisk is an inferior product to sipXecs. We felt compelled to point out the errors in his posting by responding to it in our BLOG. In preparing our response it was interesting that a number of people pointed us to prior BLOG posts where Tony had waged BLOG-war with the Nortel competitor of the day - only to be skewered by responders pointing out the inaccuracies in his assertions.
We posted our response to Tony’s blog and will share it here with our readers. So here is our view:
Presented with the disruptive threat of open source Asterisk, and the recent momentum seen in commercial channels and the enterprise, Nortel acquired sipXecs. Can you buy your way into open source credibility? Nortel’s not the first old-line company to try. Is open source a marketing bullet point for Nortel ? Its certainly not inherent behavior that’s woven into the fabric of the company!
Hey, I can’t blame them — if you were Nortel, and saw Digium and Asterisk on one side and Microsoft on the other, threatening to eat into your core business, what would you do?
So: Bravo, Nortel! Welcome to the next generation of telephony. But you’ll need to learn the strengths and limitations of what you just bought. As we have learned from our commercial customers as well as the countless numbers of Open Source Asterisk installations, SIP is not the entirety of UC. True, Asterisk isn’t a SIP proxy — because a SIP proxy alone cannot provide services the world has come to expect from phones.
To pick a specific example of the rather misleading comments in your article: It’s incorrect when you claim that all Asterisk calls go through a centralized system. We assume you’ve just been misinformed, but your claim that Asterisk is designed to always handle media streams is just incorrect. You should recognize that the Asterisk rhetoric from the sipXecs and FreeSwitch teams refers to Asterisk over 4 years and several versions ago when they last looked at the feature set. In some cases, the Pingtel/sipXecs team in particular likes to compare against specific SMB packaged systems and not native Asterisk, which would provide a more “same page” comparison. The SCS500 scales to 500 users. It certainly does not need the power already in Asterisk today.
Let’s talk real numbers. Asterisk community members have quickly created testbed platforms which process 300 SIP calls per second capacity. When RTP is managed by an Asterisk instance, we have demonstrated as many as 1900 concurrent G.711 channels on inexpensive off-the-shelf hardware. The open source versions of Asterisk were designed to handle hundreds of thousands (yes, hundreds of thousands) of end users in different circumstances, spread across multiple machines with functional role distribution in a way completely unlike a PBX. Of course, smaller SMB solutions are selling well for our Switchvox line as well as a dozen or more other companies who repackage Asterisk with limited hardware or license caps – that is their business. But please be a bit more forthcoming in your comparisons – Asterisk is being run in systems spanning huge user PBX and service delivery populations.
For sipXecs to compare themselves against the pre-packaged SMB offerings of Asterisk in different flavors is misleading. That’s like saying the Apache web server on my embedded-processor webcam can only handle 2 simultaneous streams, therefore Apache doesn’t scale on any platform. So while Asterisk can serve as a PBX, it can (and for countless customers and projects already does) also serve as an incredibly flexible service-delivery platform for UC services, custom application development, or carrier VoIP integration. It can be used on even the smallest embedded platforms (Linksys WRT54G, AA50) or on the largest voice server farms such as Integrics’ Enswitch which directly supports over 100,000 end points and over 6,000 concurrent calls just for one of their 40+ customers and integrates easily through Asterisk APIs to multiple billing solutions.
As the “wildcard” name suggests, Asterisk works with numerous protocols and codecs. When required for communication between disparate endpoints, Asterisk will intelligently negotiate and transcode between them. If the two are compatible, Asterisk will hand off the call and get out of the way. Asterisk has the flexibility of handling media if desired, but RTP between endpoints is the preferred design for larger systems, including video, which Asterisk has handled for several years now.
In contrast, the sipXecs architecture enforces the requirement that all endpoints be uniform, which pulls along all sorts of ugly forklift-upgrade requirements for businesses looking to grow into the future, or uses expensive media gateways to do what Asterisk can do in software. We can confidently say that Asterisk does UC and that sipXecs is simply a SIP platform that requires lots of other moving parts to get the job done. Don’t take our word for it—read their comparison posts, which say sipXecs needs other components to complete their system. Asterisk handles the “U” in UC (as well as the “C”) and has for some time now.
The only claim that seems to be correct in Tony Rybczynski’s post is that for large Asterisk installations, there is no comprehensive management interface for all possible aspects of the system. There are multiple web-based interfaces available for small/medium enterprise PBX-style installations - FreePBX is the most popular open source tool, and Digium’s Switchvox being an excellent representative of a commercial packaging. Both of these examples include automatic phone configuration and provisioning. However, Digium has found that large enterprise developers who wish to harness the true power of the system typically want to have call control at a much more fundamental level than what a GUI typically offers or what a vendor might consider “simplified.” Therefore, Asterisk is available as a telephony toolkit – a suite of programs and fundamental tools that allows a developer to quickly deploy new voice apps or extend existing legacy platforms if they so choose – it is as flexible as the circumstances require.
One final observation counterpoint to a comment you made: As the progenitors of the venerable DMS-100, Nortel should know by now that the age of code — or lines of source — mean nothing when compared against other software. Do more lines of code indicate more features or quality? Do fewer lines evidence efficient, bug-free code? Lines of code are typically irrelevant in doing anything other than measuring platforms against themselves over time or measuring individual coding productivity.
It’s going to take more than this acquisition to, as Tony says, solidify Nortel’s “leadership in the global open source ecosystem.” We hope that this purchase creates a more viable and useful application that can be used by the open source community – we hope this isn’t a repeat of the Vovida(Vocal)/Cisco purchase and subsequent smothering. But for the sake of the open source telephony movement that Digium started, Nortel, we welcome you to the open source revolution.
After this blog was posted, one of our managers pointed out that as of Asterisk 1.6.0, Nortel’s IP phones (the i2004 and even the i2050 softphone) can be used *DIRECTLY* with Asterisk, without the need for any protocol gateway. This is not true with sipXecs or any other SIP-based call handling engine.
Our manager’s quote, “I know, this is slightly tongue-in-cheek, but it’s a very valid difference between sipXecs and Asterisk; their SIP-centricity is not helpful to the public at large and results in extra costs” thus Asterisk stands as the firm global leader!
I didn’t know open source projects had to compete. They are all free aren’t they? I think that’s the whole problem here. Calling my list of valid issues with Asterisk rhetoric, won’t make them go away. Pretending Asterisk does not suffer from any problems and only pointing out it’s strengths is not the way to make it better.
Please admit that I have done more than most people are willing to do for completely FREE to try to make Asterisk better for several years. I only know enough to itemize the issues from that *long* experience as a an Asterisk developer.
I, in fact, invented the whole idea of the “function variables” that now are rampant in Asterisk 1.6 and there are *plenty* more things I could list if I wanted to. I also see plenty of ideas we have already implemented in FreeSWITCH starting to crop up in 1.6 as well. This is the nature of open source. If Digium chooses to actually cooperate with the open source telephony community there is much to be gained for all.
If Bill Miller sees himself as part of this open source revolution, at the most you are on the outside. You work for a company named Digium who is in the thick of it, but you yourself and new to the game. You arrogantly and wrongly include yourself as part of the telephony revolution- more perspiration and less admiration is required to join the club Bill. You are looking like a fool.
Octavian.
To the folks who were awaiting comments to be posted I apologize as they fell through the cracks for several days and this task went overlooked!
We appreciate you taking the time to comment even if they are in conflict with our opinions. that is what makes our world as interesting as it has become!
I will reply to both comments here: I see comments made about about me personally. This post from us under my name is a collection of many comments from people involved with Asterisk for nearly the life of the project. All the open source projects are free and everyone can try them all. Isn’t the world a great place that you can try all open source choices available to you without an evaluation PO?!?!?!!?
As for the comments about trashing Asterisk, the gauntlet was thrown by Tony Rybzcynski’s blog about Asterisk being inferior first. Digium has the right to defend itself and share our view. I have no issues with competitive comparisons if they are accurate, but the comparisons of all the distributions and turn key products are all incorrect and out of date as referenced on the public sipXecs site. I’ve known Tony R. for years (and have always enjoyed being on panels with him) and am pretty positive he has little open source background so he is being told or reading this information.
Anthony, I am quite aware you were a major contributor to Asterisk for years and Mark and the community certainly appreciated (and the patches you still have out there) your contributions. But you have been focused on Freeswitch for some time and things have changed under the Asterisk hood. Sure, it needs some improvements but what software doesn’t? I have written software for over 35 years (not the last few) and all software can always be improved. There have been 420 contributors to Asterisk in 2008 alone so there is plenty of core development and bug fixing taking place.
As for Octavian I am not sure what you mean. I am now nearly 3 years into Asterisk albeit not a developer. I am a spokesman for open source, open source telephony, the disruption of what all open source projects do for the good of the world, and I am an executive at Digium. Every decision is looked at with the open source community in mind, and today with John Todd at Digium working closely with me and my product management team we spend countless hours on projects that are open source community focused, some that you will see rolled out shortly. We fund other projects besides our own and work hard to fund Asterisk User Groups worldwide. I do agree I have not sweat over config files, APIs, complex Asterisk code, but I am quite familiar with the challenges people have had in the past. I would argue your comments about me as I work day and night for the open source cause.
One major agreement with Anthony M: the market is huge for VoIP, PBX, Telephony applications, and UC in all markets; as there is room today for Cisco, Avaya, Nortel, Shoretel and Siemens and many others, there is lots of room for both our projects.
I’d like to again thank you both for taking time to post and share your comments.
Digium/Asterisk is sad, go FreeSWITCH!
Bill,
Its awesome to see that the moderators just fell behind. I was crossing my fingers all along.
John Todd’s attendance of ClueCon was a great first step in Asterisk working together more with other projects. We have not seen anyone from Asterisk since Mark himself appeared at the 2005 premier. I know not much was said about me or my project personally in this posting but I felt sorry for sipX because I have a lot of respect for SIPfoundry and I think anyone who takes on a SIP based project deserves all of our praise. Lets all work towards making a respectable reputation for Open Source VoIP.
FS guy,
We thank you for your feedback even if it’s not exactly positive. It’s obviously something you care about since you took the time to post a comment. Hopefully we can work towards changing your mind as Asterisk continues to evolve.
Hi all!
and i just want to make a suggestion.
It is really an interesting article, an interesting “war”
First of all, i am not an Asterisk fan, but an VoIP fan, a field that interesting me a lot! I used Asterisk for a long time and i am also wandering on my mind that whether what i am using is the best, as there is a lot of other solutions out there, you know, openser (now is kamailio and opensips), FreeSwitch, Sipxecs, yate! And i just tried FOR A LITTLE OF TIME, so my ideas here maybe not true, please correct me if i am wrong:
1) I did visit Tony Rybcynski blog, and as a user of Asterisk system for more than 3 years, i have to claim that “Asterisk is architected to switch audio streams with all media going through a centralized system” and some other things are wrong, what he said on his blog IS WRONG.. So please check it more carefully before saying somethings not good about others..
2) I try to use many VoIP system, and i have to admit that, as a user and dev, Asterisk is a lot friendlier to use than others
3) The last and the most importance I see… Until now, through VoIP market, i see that Asterisk and Openser interaction (for large scale business) is still more prefer.. You can look through outsource sites, and search for Asterisk and openser keyword, it is found a lot more than others.. I think the market desire proves what is better
For a better VoIP world
Thank you!
Duc Viet To Ho
Duc Viet To Ho,
Thank you for taking your time to respond. We are pleased you are a VoIP fan and follow Asterisk. If I read your response correctly, you agree with what I posted in response to Tony Rybcynski’s blog. There appears to be some misunderstandings about how Asterisk is architected and how it works today. We at Digium do plan on doing more public education in the near future which should help alleviate these misconceptions. There are always enhancements being developed for software products and Asterisk is no exception.
We are quite aware that OpenSER, now renamed and forked is installed worldwide as a SIP Proxy for Asterisk which is a B2BUA (Back to back User Agent for those not familiar) and we see this at many of our largest service provider partners. We are working on a case study now that utilizes the Asterisk-OpenSER combo along with MySQL as the data base. Asterisk is the call control/IP PBX/feature server which it does quite well.
Keep your eyes open, we have lots of good Asterisk-related happenings. Again, thanks for the post!
….bill
[...] reaf what I read, The Hyperconnected Enterpriseby Tony Rybcynski from Nortel, on TMC Bloggers. Beelinebill told me about it. Tags: Asterisk, SIPfoundry, Nortel, TMC Bloggers, Beelinebill Snap Voip [...]
Can disruption be coming to Digium’s commercial offerings?
Speaking of Nortel…now that they have aquired Pingtel, and have released the SCS500 SIPFoundry based IP-PBX platform for SMB, I suspect that Digium|Asterisk|SwitchVox will need to provide some more information as to how they will compete over Nortel|Pingtel’s offerings in the commercial marketplace.
Point is what is it that pingtel has and nortel does not … in terms of technology.
Nortel’s stratagy in choosing SIPFoundry (as stated on The Hyperconnected Enterprise)
will cause a lot of confusion for our customers moving toward adopting open source based communications and I feel that that it would be nice to have some answers from Digium that provide a clear direction for customers wanting to buy the Digium solutions. Nortel is making some stong points about SIPFoundry being a better platform and are touting their extensive developer community, with more than 60 developer partners and close to 400 active members.
How will Digium answer this: “If your career is on the line when the phone system fails and you have nobody to call have a look at the commercial version of sipXecs, called SIPxchange, and provided by Pingtel Corp … Pingtel offers full commercial support as you would expect from any commercial company.” That said Nortel’s not been shy about their support for open source and their use of open source. As for Digium, their Asterisk technology and brand is the biggest name in open source VoIP.
Lastly Fonality has scored another $12 million in financing from Intel Capital and Draper Fisher Jurvetson — and some of the money could be earmarked for acquisitions. Impressive, but what does it mean for VARs and solutions providers need ing to choose a solution?
So let the battle begin.
Moe Schwartz- Technical Director Westcon
“Since it is an open source project called sipXecs they already had the code, so why bother? We find this intriguing.”
Considering the liberal LGPL license under which sipXecs is licensed, I can only venture that Nortel was interested in retaining the core developers of the project. It is an interesting point however, because the licensing model is one of the main differences between Asterisk and sipX that has not been discussed here.
Digium’s past insistence on retaining the rights to any contributions made to Asterisk makes it impossible to include GPL code from other projects, and often time is spent on re-implementing solutions when better code is already available (SpanDSP, GUI, SIP stacks, H.323). I hope Nortel’s increased involvement with sipXecs will give Digium cause for reconsideration of it’s development model (GPL only, 6 month release cycles, one GUI).
Another big difference is the end-user experience. Asterisk is a very flexible piece of software, but it did not arrive to where it is today overnight. Early on, providing a GUI for common administration tasks was seen as an integrator/reseller niche. By the time Digium realized that a GUI is essential for Asterisk to gain broader acceptance, a dozen competing solutions had already sprung up. Now we are left with quite a few popular web interfaces to administrate Asterisk, and no clear winner. Even Digium is pushing forward two entirely different solutions.
In general, choice is good, but this kind of lack of direction makes it harder for companies to develop a support model around Asterisk. Basic tasks, such as adding a phone or terminating a SIP trunk are handled completely different in Asterisk NOW, FreePBX, Fonality PBxtra, SwitchVox or Trixbox.
The sipXecs GUI on the other hand is both consistent and well-documented, but also highlights a fairly spartan set of features in some areas (most notably in voicemail).
It’s been exciting to witness the acceleration of Asterisk over the past few years, and finally seeing sipX reach it potential is cause for excitement as well.
Asterisk and AMP (Freepbx ~ Asterisk@Home and Trixbox) got us to where we are today.
The fact that ANYONE could download an ISO and have a working box in under a day (well maybe not anyone, but a lot of folks who would never had done it without the ISO ease of load).
AAH got Voip / Asterisk a “hot geeky thing to do”, then Trixbox got everyone else who could spell Voip to try it.
Now look at us we try anything and everything because there is no barrier to what we can dream up.
We know what ever we want to do, that it CAN be done, we just need to figure out HOW to do it.
Asterisk has it flaws and yes the changes from 1.2 to 1.6 are great; things are getting better (faster / less CPU on reload) but all in all Asterisk has been a stable usable product for the last few years.
FS also has it’s flaws; no GUI / XML coding is hard to read for us almost blind people.
And it’s strong points like as a conference server it wins hands down, so you guys get together, and work it out, I will by the beer if need be.
The point is to take the customer from the OTHER guy, not from one of us.
As the leading installer & integrator of Asterisk (and other VoIP softwares) in Oregon, we love to see the projects use what works, drop what does not, share best practices, and end up with the best possible solution. We do not care who is who & what is what, as long as it works, works very well, is well engineered, and as bullet-proof as possible. We fight eery day with bad hardware, bad software, bad IT support, user’s expectations, and on down the list. (as I am sure lots of you do too) We thank every single developer who has ever contributed any code to any of the projects mentioned on this page & in fact thank them in advance for any future code they improve inside any of the projects mentioned on this page. Speaking as someone who makes a living selling chunks of his brain & chunks of his time, I hope that each & every project inches forward. The competition between them is wonderful. I do not think of these things as wars, and one project (or person) winning. In fact, I think those who might think that way are missing the point. You can win. There is no prize.
(If it works, we will install it, if it does not, we will not, period.)
Thanks every one. Keep the good stuff coming.
Guys,
I prefer to think in terms of benefits for the entire VoIP community.
Mark Spencer and their guys, take to the top the IP PBX as a commodity, and we must appreciate it. Thanks to it fact, I am learning day by day about this wonderfull world.
Asterisk contribute to the VoIP community with many improvements, for example, creation of IAX and IAX2 protocol to improve communication between phones across firewalls, in a better way than SIP. So good than many chinese manufactures, adopt IAX2 as another protocol in their IP phones.
Anthony Minnesale is a great architect and developer. He contributed with FS to the community with a improved and a very well product. As a consecuence, we have another OS product, with diffrerent architecture, flexible, robust, scalable, and sure will be a place to drive in a better way, common facing problems in softswitch and pbx products of today.
Another product ? another opportunity to change the way … another focus to the same problem: human communication. Competition ? is better !! is a great motivation to make the things in different ways, in our eternal search of Perfection …
I just want to thanks to the geeks, genius and every motivated people in this industry, and to say: “guys, don’t argue please, just do what you better know to do … better tools !!”