Microsoft Teams, Direct Routing and SBCs (pt.1)
My journey creating a scalable SBC as a Service for Microsoft Teams Direct Routing is over at Snapsonic.com.
So it begins. Skype for Asterisk falls.
It looks like the first victim in the Microsoft acquisition of Skype is Digium and the open source PBX – Asterisk. The following is an email sent to existing Skype for Asterisk users…
Skype for Asterisk will not be available for sale or activation after July 26, 2011.
Skype for Asterisk was developed by Digium in cooperation with Skype. It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software. Therefore Skype for Asterisk sales and activations will cease on July 26, 2011.
This change should not affect any existing users of Skype for Asterisk. Representatives of Skype have assured us that they will continue to support and maintain the Skype for Asterisk software for a period of two years thereafter, as specified in the agreement with Digium. We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion.
Skype for Asterisk remains for sale and activation until July 26, 2011. Please complete any purchases and activations before that date.
Thank you for your business.
Digium Product Management
One has to wonder what will become of Skype Connect, Skype’s answer to SIP Trunking. Will Microsoft shut off the Skype Connect vendors (Cisco, Avaya, Grandstream, etc.) as well?
Original forum post here.
VoIP Network Monitor 'SIPQOS' Launches Beta
Many of us have struggled with VoIP Network Monitoring, keeping tabs on our network without having to manually review the health is always a hassle and concern. For every network my team erected we needed to erect a proper monitor. For smaller networks and even VoIP phone systems the traditional Network Monitors were far to expensive to implement and required port mirroring which meant servers had to be deployed in the VoIP network that required monitoring.
So, we created SIPQOS… SIPQOS is a service that allows VoIP network administrators to attach virtual SIP endpoints to their network which send calls to-and-fro and monitors those calls for interruption. It’s a simplistic approach to a complex problem, if the network drops a registration or if a call fails it’s likely (from personal experience at least) that the issue applies to the entire network and other endpoints are experiencing the same problem. SIPQOS won’t take the place of more expensive in-network solutions but it does a great job of providing redundant VoIP network monitoring and SIP-based VoIP phone system monitoring as well.
An excerpt from the announcement we made on the 10th…
VANCOUVER, February. 10 – SIPQOS (pronounced SIP-KWOSS), a new entrant in the VoIP network monitoring market has launched a beta of its remote VoIP network monitoring service today. SIPQOS released the first product to bring the power of remote VoIP network monitoring by combining embedded SIP (Session Initiation Protocol) User Agents, web services and some secret sauce. SIPQOS monitors VoIP networks remotely and alerts network administrators when a problem has been detected.
SIPQOS is doing a great job for us and provides redundant VoIP network monitoring on a production network we run today. It also fills a void where others solutions fell flat, SMS alerts are critical and SIPQOS delivers in spades on that front. Those interested should give it a whirl, it’s free to sign up and the plans after the 30 day trial are cheap by anyone’s standards.
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.
WebRTC.is 