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.
iPhone 4G, Data only + VoIP, Google Nexus One coming to Canada?
Image from PC World
Ok, so VoIP over 3G isn’t quite there, but 4G is not far off.
It would seem that Apple believes 4G is ready for voice and video calling in Korea at least. According to a Korean blog, Korea Telecom will be deploying the iPhone 4G in June of this year. The new device will sport forward and rearward facing (5-megapixel) cameras, an OLED screen and a video calling service.
It occurs to me that with all that is going on in the mobile space, at least one of the providers would have come to market with a data only + VoIP offer. Well, there is still a chance that might happen, in Canada. If we look at the recent spectrum auction it is plain to see the potential players who could bring the Google Nexus One (N1) to market in Canada. It seems that there are only 2 possibilities; DAVE wireless or Wynd Mobile.
Since Wynd has launched there has been no mention of the N1, so maybe it’s DAVE wireless that is bringing the N1 to market in Canada? Will we see a data only offer? One can only hope.
I am an iPhone 3GS user now, but I would jump ship in a heartbeat if I could get decent coverage at a decent price with 3.5/4G + VoIP service of my choice. This seems like such a no-brainer and could seriously disrupt the industry. Let’s get on with it already!
The year VoIP came back from the grave.
update: FCC sees VoIP as the future.
I don’t want to go on the cart!
Some of you may remember rumblings in the blogosphere, “VoIP died or VoIP is dead” around this time last year. Whatever the context, I think it should be clear by now the VoIP is not dead, nor dying. As a matter of fact, VoIP has never been less dead.
Some may argue that I am taking some of those statements out of context. Semantics. Some said “buddy list” centric calling is the future, hence VoIP is dead, again – semantics.
Call it what you like, VoIP is here to stay, Mobile VoIP is only just getting started.
Give it 5-10 years (not long considering the PSTN has been around for more than 100 years) and everything will be * over IP, including Voice and Video.
Will Rogers follow AT&T's lead and allow VoIP over 3G? Yes.
Yes, they will.
Jailbroken iPhone running VoIP over 3G on Rogers
1. Rogers has cornered the GSM market in Canada and is the only carrier to offer the iPhone, but that is about to change. Telus and Bell have tag-teamed to erect an HSPA+ network and will be offering the iPhone as early as next month. Just in time for the holiday season and with plenty of time to ready themselves for the 2010 games in Vancouver.
It’s true that 3G is not yet ubiquitous which mean VoIP over 3G is not something that will drive massive adoption in the near term, but it will be enough of a detractor for a good percentage of the users to not choose Rogers if Telus and Bell allow VoIP over 3G on the iPhone.
2. Rumors have it that Globalive / Wind Mobile is hot on trail of Rogers and will be completing Phase 1 of their network build-out as early as this spring. They too might be carrying the iPhone. None of the big three want to get beat out by the new guy on the block.
3. Other devices on the Rogers network already have apps that deliver VoIP over 3G service. It’s not the network that is the limiting factor here, it’s the Apple app store and the contract they have with the carriers representing the iPhone.
4. Net Neutrality. I am sure that Rogers would like to avoid getting dragged into the same kind of kerfuffle the FCC has been crowing about in the US. The Internet does not stop at the desktop, so why should those it be left out of such conversations, it simply shouldn’t.
It’s should also be clear that Apple would prefer it if the carriers would allow VoIP over 3G. It would mean more devices sold and more interesting apps in the app store. I just can;t see Apple saying “no thanks” to VoIP related (product and service) revenue in the app store.
I think the question is more a matter of ‘when’ as opposed to ‘if’. Hopefully it’s soon!
iPhone Video to launch at WWDC?
Update: Yes, it was indeed launched and it’s called the iPhone 3G S but no video calling as yet.
There are rumors abound regarding the next release of the iPhone, every tech blog known to man is all over this like a fat kid on a smarty.
The iPhone 3.0 SDK has pretty much been proven to support video so a iPhone Video product seems to make sense. What kind of video? Recording full frame video is one thing but transporting that over 3G is quite another. My guess is it will not support real-time streaming or video calling on 3G, the question is will it deliver the goods on WiFi.
It will be interesting to see what happens at WWDC (running from the 8th to the 12th), the new iPhone is sure to launch at this event.
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.
How much bandwidth do I need for Response Point? G.711 vs. G.729
G.711 is the default audio CODEC for most Response Point phones and requires approximately 90Kbps bandwidth upstream (your voice going out) and 90Kbps bandwidth downstream (your caller’s voice coming in).
To calculate peak usage take the peak concurrent callers x 90Kbps. For example: 5 concurrent calls x 90Kbps = 450Kbps is the required bandwidth for each direction. Keep in mind, this does not account for VPN usage for remote users or voice mail to email etc.
As an example, if you have a 1Mbps ADSL connection from your service provider, you might have an upstream bandwidth of approximately 700 Kbps. A conservative approach is to estimate just over half of the upstream bandwidth is available, ISPs generally over-sell their bandwidth. In this case, you could safely support 4 simultaneous G.711 calls if you were not doing anything else (e.g. downloading email, listening to online radio, downloading large files, etc.) on that connection.
The SMB Digital Voice network also supports G.729, which uses approximately 20Kbps bandwidth upstream (your voice going out) and 20Kbps bandwidth downstream (your caller’s voice coming in) for each call. G.729 provides very good call quality while minimizing bandwidth usage. The only noticeable difference would likely arise during on-net calls (calling other users on the SMB Phone network). G.711 offers a higher quality on-net call because G.711 does not compress audio, but as soon as the the call is handed off to the PSTN the call quality between G.711 and G.729 is hardly noticeable.
G.729 offers some real benefits, the most obvious is the 400% decrease in bandwidth capacity requirements. G.729 also handles Jitter more efficiency during times where low bandwidth / high congestion would likely render a similar call using G.711 unintelligible.
You can force your phone to use G.729 on Response Point handsets but some are harder to configure than others. For example, on Aastra 675x phones the global SIP settings are grayed out out via Javascript on page load making it tough to set the codec.
As a general rule of thumb, we like to recommend an independent broadband connection that you can use for Response Point. You may want to acquire a router that has dual WAN link failover, VPN Server (for remote sites) and some QOS traffic shaping functionality.
Response Point VPNs and Remote Workers
I wrote an article over at the SMB Phone blog on Response Point VPNs and remote workers. If you are having some issues with VPNs and Response Point this might help.
Skype for SIP, it's about time!
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Back in 2004 I wrote a post relating to the VON Canada Panel I sat on with Niklas Zennstrom. It was an interesting debate on open standards (SIP in this case) and closed networks, specifically Skype. I was quite vocal about how silly I thought Skype was not to include SIP, a few of you picked up on that 😉
It looks like something good came of the eBay purchase as we now see a Skype pushing towards open standards, good stuff!
On a similar note, I heard a rumour that it’s likely Jason Fischl the current CTO at Counterpath (Xten) will be going over to work with Jonathan Christensen (General Manager – Media Platform) at Skype. Jason was an early advocate of SIP in the IETF and works with some of the best minds on the subject: Cullen Jennings, Robert Sparks, Alan Duric come to mind.
This could get interesting.
I will do some testing with SkypeforSIP & Response Point and post the results along with my comments on what this new offer from Skype might mean for Response Point.
WebRTC.is 