Skype's email to me re: Skype for Asterisk
I just received this email from Skype’s PR firm…
Hi Erik,
Here is Skype’s official comment regarding Skype for Asterisk. You can attribute this to Jennifer Caukin, spokeswoman for Skype.
“Skype made the decision to retire Skype for Asterisk several months ago, as we have prioritized our focus around implementing the IETF SIP standard in our Skype Connect solution. SIP enjoys the broadest support of any of the available signaling alternatives by business communications equipment vendors, including Digium. By supporting SIP in favor of alternatives, we maximize our resources and continue to reinforce our commitment to delivering Skype on key platforms where we can meet the broadest customer demand.”
Thank you,
Cassie
Call me crazy but if I have to pay to integrate Skype into my phone system, where I already have a phone service that I am happy with, why would I do that? Maybe I just want to be able to make/receive Skype calls on my SIP-enabled desk phone? If it doesn’t hit the PSTN why do I have to pay? Seems like an odd approach for a company that has a long history of working around POTS, much to the delight of their users.
Integration with SIP is great, don’t get me wrong, but it would be nice if Skype talked SIP and was ‘still’ free. Seems like a massive oversight on behalf of Skype or am I missing something?
Asterisk + Skype + SMB = Freetalk Connect
Jazinga and Freetalk have combined efforts and the result is a Skype enabled SMB phone system called Freetalk Connect.
The press release:
FREETALK Partners With Jazinga To Create FREETALK® Connect
Companies Collaborate On Skype-enabled Small Business Communication System
Featuring Set Up In Less Than 15 MinutesMIAMI, January 20, 2010 — As the result of a new partnership announced today at ITEXPO East 2010, FREETALK and Jazinga have created the FREETALK® Connect, a full-featured unified communications system that is the first to feature Skype for SIP and Skype for Asterisk functionality.
FREETALK and Jazinga collaborated in designing the FREETALK Connect, featuring a do-it-yourself (DIY) technology approach that can be configured in less than 15 minutes, enabling users who are not tech savvy to use it without formal training. This new class of DIY communications system allows anyone with basic knowledge of computers to install and maintain the office phone system. SIP, Skype and traditional PSTN phones can be plugged into the network, and the FREETALK Connect auto-detects and configures them. An onscreen wizard guides the user through setup. Adding users and administering the system after install is equally simple.
Further distinguishing the FREETALK Connect is its intelligent routing capabilities. Incoming Skype calls, as well as SIP, PSTN and IAX2 calls, can be routed to any local or remote Skype user, SIP, analog or mobile phone. Additionally, the FREETALK Connect enables users to set up “Find Me, Follow Me” features, and provides a unified mail box that consolidates messages from voice mail and email into one mailbox.
Some of the key features from the Jazinga platform found in the FREETALK Connect include:
Callback / Dial-around
Access to Skype Buddy lists
Auto Attendant / IVR
Paging
Call Parking
Remote Extensions
Music on Hold
ConferencingThe FREETALK Connect also has an easily configured and updated:
Managing routes to users, telephone services, and applications
Providing SIP/Skype telephone service management
Router management (networking, port forwarding, DNS, DHCP)“Jazinga’s products consistently ensure call integrity by integrating quality of service and prioritizing voice traffic on the network into an affordable, simple product,” said In Store Solutions COO Craig Smith. “There was no question that FREETALK wanted to partner with Jazinga to develop the FREETALK Connect, because it continues our goal of working with the best providers to distribute outstanding products around the world.”
“FREETALK Connect is designed for small businesses with between 2 and 49 users, an undersold market that desperately needs UC functionality,” said Randy Busch, CEO of Jazinga Inc. “As a result of our partnership with In Store Solutions, the telecom technology playing field is much more level between larger enterprises and their smaller competitors.”The the FREETALK Connect is marketed through Skype Shop, which is operated by In Store Solutions. The unit initially will be available to registered U.S. Skype users beginning in March.
For more information about FREETALK Connect PBX or to order a unit, visit
About FREETALK
FREETALK is a product innovation catalyst – identifying market gaps and working with its global partners to design, manufacture and quickly bring to market products that disrupt traditional categories. Leveraging untapped market opportunities, FREETALK products are designed to be environmentally friendly, sold online and delivered globally at aggressive price-points. Always at the forefront of innovation, FREETALK is known for creating synergistic products that add unique value to its partners’ branded points-of-sale.
About Jazinga
Jazinga Inc. develops communications products for small businesses and homes. The Jazinga system provides enterprise telephony and data functionality for this market, but at a fraction of the cost and without the setup complexity of an enterprise-class IP PBX. Jazinga Inc. is privately held and headquartered in Toronto, Canada. Additional information is available at http://www.jazinga.com.
Contact:
Sue Huss, for In Store Solutions
sue.huss@comunicano.com
+1 619-379-4396
Jazinga came to market a while back with a Asterisk appliance that is not much different than other you would find in the Asterisk market today. Skype recently announced their Skype SIP Trunking capability which is helping Skype become more open standards compliant, paving the way for deals like this one.
Since I have not tested the system myself I can only speculate that it is not huge departure from other Asterisk systems, which are not trivial to set up. Let’s hope they did their homework and come to market (March) with something that is much less technical and more end-user friendly, like Response Point.. was.
One thing that I find interesting is that it will be sold via the Skype store to US registered Skype users. If you were wondering what the connection is between Freetalk and Skype; the creators of Freetalk are also the curators of the Skype store. Ya, you heard me right. The company that created Freetalk (In Store Solutions) operates the Skype store. Which makes one wonder if there is overlapping ownership between Skype and In Store Solutions.
Something else that I find interesting, and not just because I am one of the founders of Xten/Counterpath, is how this announcement relates the recent announcement of the Asterisk/Digium softphone from Counterpath. Which may be why In Store Solutions decided not to leverage the Digium or Asterisk brand in this release, maybe they see the new Asterisk Bria softphone as a competitor in this instance?
I expect this will not be the last Asterisk-based phone system to incorporate Skype functionality this year, but it would seem as though they are the first, congrats to fellow Canadians at Jazinga.
SIP Trunking and Hosted PBX in Canada will speed HD Voice for small business
SIP trunks are simply another way of saying VoIP Provider for your phone system. A SIP trunk is a connection from a PBX (phone system) using SIP (Session Initiation Protocol) to an ITSP (Internet Telephony Service Provider).
It might sound complicated but it’s really quite simple, SIP trunks take the place of your legacy telephone company. Most phone systems out there today are more than a couple of years old and are likely based on circuit switched technology. Newer IP-PBXs use packet switching technology, which means they leverage the Internet to deliver the same features you have now, and then some. The difference could be minor or major depending on what your PBX is capable of and what your ITSP can deliver in terms of features and functionality.
Since the PSTN (public switch telephone network) is tied to aging circuit switched technology it has limitations in terms of what media it can support. Essentially, it can deliver low quality voice, that’s it.
SIP Trunks replace older PRI and POTS interfaces that we are used to and bring to the table a wide variety of communications options. Depending on your IP-PBX and your ITSP you could potentially look forward to HD (High Defenition) Voice and potentially HD Video.
HD voice (and video) for small business in Canada will happen, it’s only a matter of time. As broadband providers increase upstream bandwidth and dual WAN link-failover devices become common place, SIP trunking will accellerate in growth and on-net (calls made on the ITSP network) HD Voice will become common place.
Unfortunately, HD communication off-net (eg. PSTN) is not going anywhere at any great speed. Jeff Pulver is back as he reboots the communications industry with his new HD Communication Summit. I welcome Jeff back with open arms, if anyone can convince operators to increase speed towards wide-band/HD adoption it would Jeff Pulver.
Today we can see SIP trunking providers and hosted pbx providers supporting wideband codecs and devices on their networks. This will allow user to communicate in high definition with other users that have devices that support it, in brief you could have better calls between you and your colleagues in the office and remote office workers connected to the same PBX, and that is a step in the right direction.
Ribbit vs. Lypp
VS 
I have had a few people ask me to describe the differences between Ribbit and the Lypp API.
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UPDATE: Ike Elliot has some good points about the un-evolution of VoIP
The one thing I might say to Ike is, "you're right, in more ways than one". VoIP has not really come all that far and sometime it complicates life more than it needs to. I think I can help you in one way though Ike, check back in a week and you will see what I mean.
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UPDATE: Garrett Smith adds some food for thought
Garrett mentions "Lypp appears to be a solution for mobile professionals that aggregates AIM / AOL, Google Talk / Jabber, iChat MSN and Yahoo! Messenger contacts and allows for group or conference calling via your cellular handset. It also does not leverage the IP network, in favor of the wireless network and or PSTN." I can see why Garrett would think that, the current site says nothing about our Next Generation Conference Calling service, VoIP API or Rails plugin. Keep your ear to the Rails Garrett, that is soon to change 🙂
As a developer Garrett had some comments on the APIs. Garrett mentions that he could not really use either API which I found a little disconcerting. Our goal is to make sure that anyone who understands XML or Rails can use this API. The Lypp API is published here: lypp.com/api and can be accessed by simply sending an email to api@lypp.com requesting a key.
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UPDATE: Luca Filigheddu with some thoughts of differentiation
Luca makes a good point here about the importance of differentiation.
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UPDATE: Moshe Maeir makes a great anaolgy.
Yes, you are correct Moshe. We are bootstrapping this venture and our poultry investment over the pat year is lunch money when compared to what Ribbit has raised but I think I would still prefer to be driving a Chevy 🙂
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UPDATE: Thomas Howe reflects on the differences and makes some good points.
Thomas is a smart guy and I have a great of respect for what he is doing in the voip mashup space and what he has done in the past. His comments on my initial post are well taken. On the last comment, I am not opposed to softphones, not at all. It's just that I have seen softphones deployed in almost every scenario imaginable and the take rate in the business community has been low. Mostly due to technical network issues like double firewalls and zero-tolerance VPNs. All that aside, I am very positive about the future of softphones and firmly believe you will see one in the Lypp lign-up, when the time is right.
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UPDATE: Andy chimes in by ringing the bell. <ugh>
I think Andy might have slightly missinterpreted my intentions when writing this post but hey, a little spice never hurt anyone 😉
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First let me begin by saying I know Ted Griggs and I respect him greatly, he has a great track record for building innovative companies that push the boundaries of technology and communications.
I was the initial designer, sales guy, visionary, president, co-founder and COO at Xten (Counterpath) which since inception has dominated the SIP softphone SDK space. In other words, I think I may know a thing or two about building softphones.
Fyi, Ted and I will be presenting on behalf of our respective companies at Wireless Innovations in April.
With that out of the way, here is why, when I started down this path, I did not choose to reinvent the softphone at the edge of the network.
The edge of the network is a nasty place. Bandwidth issues, carrier packet shaping, lack of end user control and costly redundancy solutions make it nearly impossible to deliver a predictable and reliable telephony service.
Much like turning on the lights when you get to your office, that phone on your desk had better work as expected.
In saying that many professionals use Skype and other softphones, like X-PRO, X-Lite, eyeBeam etc to make calls over the net everyday. But you can bet when it comes time to make the calls that really matter they are not using a softphone on the open Internet, at least not after it suffers major packet loss more than once during a call of significance.
This is also why traditional telephony will be around for decades to come. The PSTN still rules the roost. Setting aside for a moment the unwillingness of the carriers to allow other providers to simply stand up a service that will cannibalize their revenues, reliability and Quality of Service (QoS) is still a major issue.
At Gaboogie we steered away from the softphone or using any VoIP at the edge of the network in our initial plans. We made that decision early on because we believe VoIP at the edge is still not ready for prime time. If you don’t believe you obviously have not tried a best efforts VoIP service in Canada. I have not found a single best efforts offering that does not drops calls, drop packets and well… just generally suck.
So what is Lypp then?
The Lypp API was built to support advanced conferencing and was meant for critical calls for companies that require a dependable service. That does not mean a developer could not use it for more typcial telephony integration, which in fact some are already doing. Using the API directly via XML or by way of the Ruby on Rails plugin developers can add traditinoal telephony and/or conferencing capabilities to their apps in as little as a couple of hours.
We have constructed a very robust network that is redundant and dynamically scalable to handle billions of minutes of call volume per month. Our call back methodology (been around forever) keeps the VoIP in the core of the network. If your landline or cell phone is on, so is our service. Our customers do not suffer from call quality or reliability issues in the same way best effort VoIP service users might.
Developers leveraging the Lypp API can expect a higher degree of call reliability and call quality, more of the time, than any other best efforts VoIP service in North America, period.
Best efforts VoIP, whether you are using a Polycom VoIP handset and an Asterisk PBX or you are using a Ribbit inspired softphone, will likely not match up with the reliability you have come to expect from the legacy telephone networks. However the feature set of POTS (Plain Old Telephone Service) pales in comparison to what VoIP can offer.
Some day we will have the kind of IP infrastructure that will make the edge of the network near bullet proof, but in my humble opinion, we are still a ways off. When we do get there Gaboogie will be ready to leverage its SIP network to the absolute maximum.
WebRTC.is 