Source: snapvoip.blogspot.com

I read about backup for VoIP on a leading Magazine today. One of the backups it suggest is that to keep the good old copper wires, PSTN, and that is what I am doing, at office, home office and home.
They also suggested and I also have UPS backups fro my routers, DSL routers and VoIP switches.
Our Asterisk Servers are protected and our VoIP Phones, a mixture of Cisco, Linksys, Astra, Avaya and Grandstream phones also protected by UPS, over POE.
I even have a Solar Power system at Home and home office. System runs well with most of the time sending excess power to the grid.
But with all these systems, I have lost VoIP service, in two incidents when we lost power in the area, we lost our phone service. But our office and home was well lit.
Why? because the central switch (Belong to service provider) that our DSL service came from also lost power. So only service we had was good old POTS service, and our cell phones, iPhones.
I do not have cable service but I am not sure if they also connect to central switches. And if they do, it will be the same picture, your area loose power, you get no VoIP.
So make sure that you have at least one pots line with a old type phone and well charged cell phone.

Source: goebel.net

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Much has been said about startups like Ribbit, Tringme or Flashphone which use the Flash browser plugin for click to call widgets. Aswath Rao even declares 2008 the Year of Flash based VoIP Clients. I can only say that you don’t have to wait till next year to call me for free using Flash. I love my Tringme call widget:

These calls are entirely free to you, because the caller speaks into the Flash widget on my website using a headset or the laptop’s built in phone and speaker. On Linux the sound is a little bit weird. The automatic voice, which says "please wait while we connect your call" before every connection, sounds like a 45 rpm record played on 33. The phone call itself sounds like Mickey Mouse, but still the the words are understandable. On Windows everything works just perfect.

Also to me these calls are entirely free. Other than my widgets from Sitfono and Voxalot where I have to pay to call the person who wants to contact me.

I achieve this by using FWD as SIP provider to power the Tringme widget. The Tringme account website says "Connect my phone and voicemail widget to Phone number or extension". Unfortunately it accepts only numbers and no SIP addresses in this input mask, but as a workaround I have simply put my FWD number there. In the "TringPhone SIP Settings" part of the account configuration I left my FWD login data. Which means that every Tringme call is in fact a free FWD on net call. You can probably do the same with Gizmo Project’s SIP account data and phone numbers, as well as with many other VoIP providers.

Also there is another widget for people who don’t want to talk to me, but just leave a voicemail.

Only seconds later I get a call and a voice says "You have a Tringme" before it plays the message. The Tringme widgets are much better than Gizmocall which also allows free calls from a website.

You could call me for free by simply typing http://www.gizmocall.com/mgoebel in your browser’s address bar. This website also uses Flash, but additionally you have to install a plugin for Windows or Mac. For ten months yet Gizmo owes us a Linux plugin. Although the company’s CEO, Michael Robertson, even has his own Linux distribution, Linspire.

But why bother? The Flash browser plugin gets more and more versatile and works on all platforms. It’s a new way to disrupt the telco industry, circumventing the PSTN and offering a new option for free phone calls that so many people appreciate.

So, if you want, please give me a Tringme call!

And, before you ask: No, I couldn’t get Truphone’s Facebook application running, which should basically do the same like Tringme, only that it uses Java. After one week of tinkering I gave up. But congratulations for winning the "Red Herring 100 Global" Award.

Source: goebel.net

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You know that I bashed Facebook very hard for being a terrible time sucker. Many Web 2.0 applications need too much attention, compared to their value. But there are some utilizations that make me smile, because the unleash the potential of Web 2.0 without wasting my precious time and money. Like Voxalot’s latest Facebook application, VoxCall for Facebook, that really disrupts telecommunications. It let’s me make free phone calls without touching the PSTN. Read the announcement:

On Monday 19th Nov 2007 Voxalot will be officially launching our new social communications application for Facebook called VoxCall.

VoxCall is an exciting new initiative from Voxalot that allows Facebook users to click on their friends and initiate phone calls. The beauty of VoxCall is that it is self-organising in that if your VoxCall friend changes their contact phone number, you don’t even have to be notified… VoxCall will use whatever number they have registered.

VoxCall also offers both public and private chat rooms where VoxCall friends can get together for a group discussion.

The underlying technology that VoxCall uses to connect calls is Voice over IP addresses (often known as SIP URIs). When you add the VoxCall application, you will be prompted to enter your SIP URI. To ensure that you are the rightful owner of that number, VoxCall will display a PIN number on the screen and then call the number you entered. Your phone will ring and you will be prompted to enter the PIN, which is validated.

As such, VoxCall supports calls between friends that belong to *any* "open" voice network (not just Voxalot).

The beauty is that VoxCall uses VoIP without touching the PSTN. My buddies just enter their SIP URI and I can call them with just one click in Facebook. When they change their SIP address I don’t have to bother to update my data since their Facebook button stays the same. We stay connected for free from SIP to SIP.

I find this much more nifty than the Facebook apps from Jajah, Jangl, Jaxtr, Rebtel, IVR Technologies, iotum, Sitfono or Grandcentral. They also connect people on Facebook and let them call me for free, in most cases. But there is always a telephone number involved, so that someone has to pay an incumbent telco which provides them.

Source: snapvoip.blogspot.com

Users can now experience the huge benefits of VoIP without the need for a PC. Hip-IP.com now stock the RTX DualPhone 3081 which allows the user to make and receive VoIP enabled telephone calls without a PC.
With the increasing popularity of the SIP based VoIP providers this provides a genuine alternative to the Skype™ based DualPhone. With the option of including upto 8 handsets in one gateway it also suitable for use in offices as well as homes.

The dual mode technology also means that these excellent products can also make regular PSTN (land line) telephone calls from the same unit. The product has easy to navigate keypads that allow you to easily distinguish between land line and Internet calls.

With its timeless Scandinavian stylish design it fits into both the modern home and office environment. With the option of up to 8 handsets (each with individual phone numbers) connected to a gateway, you can make 3 sip calls and 1 PSTN call simultaneously, and with the built in PBX functionality – the phone is a perfect work tool for smaller enterprises and it just makes every day at work a bit easier.

Add to this a sturdiness that reassures and a 1 year manufacturers warranty they are an attractive prospect for both newcomers to VoIP and old hands.

With the recent launch of the Siemens Gigaset range the market and popularity for this type of product is hotting up. Consumers are now waiting with bated breath for the next development in this technology which is bringing much lower call rates (if not free calls) to the masses. You can be sure whatever those developments are Hip-IP.com will be on the case.

Hip-IP.com is an online retailer specialising in Internet Telephony equipment including VoiP and Skype™ enabled DualPhone, VoIP Speakerphone and VoIP USB Phones. We dispatch products worldwide.

Source: snapvoip.blogspot.com

I found this application and the company while looking up Skype alternatives. Damaka claims that it’s SIP (Session Initiation Protocol) based Personal Softswitch™ application empowers the end user with multimedia communication and collaboration capabilities and that Damaka is the first company in the world to offer Softswitch technology to the end users.
According to the website, Using Damaka software, users can make phone & video calls to other Damaka users with PSTN like sound quality. Damaka offers real-time presence, IM and pc-to-pc calling, secure video with more functionality planned for roll out in the future versions of the application.

I am in the process of testing the software and features. Following is a Press release from Damaka.

RICHARDSON, Texas – damaka®, an innovative communications and collaboration software company, announces its latest P2P Personal Softswitch TM application release which would enable users to connect to SIP networks globally. As part of damaka’s effort to establish global connectivity to disparate networks and services, users will be able to connect to SIP networks by making simple configuration settings on the application. "We strongly believe that communications and collaborations should be open and seamless, while maintaining high level of security, to all users across different network boundaries", states Satish Gundabathula, CTO of damaka.

This release provides both novice and advanced users to setup SIP connectivity by selecting either simple or advanced configuration options. Enterprises worldwide can now use damaka to connect into their existing SIP infrastructure seamlessly and benefit from the feature rich environment provided by the damaka Personal Softswitch TM application. "damaka just works" with any SIP infrastructure like Cisco ® CallManager ®, Microsoft ® LCS ® and other SIP IP-PBX’s.

As part of its endeavor, damaka will continue to Interop with more and more SIP networks in the marketplace to create seamless interconnectivity. IMS connectivity is also part of this endeavor. IMS service providers can now avail of damaka software as it is SIP and IMS compliant to deliver their features.

Source: snapvoip.blogspot.com

Thomson (Euronext 18453; NYSE; TMS) today announced that more than 6 million VoIP telephony lines deployed by operators around the world are managed by Cirpack Class-5* voice switches from Thomson. This represents a 50% growth of users in just six months. Class 5 voice switches allow operators to provide primary line telephony with emergency numbers and legal intercept with advanced voice features such as call forwarding and call conferencing to their customers.

With the launch of its solutions for fixed mobile convergence in 2006, clients for the Cirpack switch now include large mobile operators such as SFR, as well as cable operators, ISPs and telcos. The Thomson’s Cirpack voice switching platform is being used by over 85 operators and ISPs in over 35 countries.

“Voice over IP is now a wide phenomenon. In France, for instance, about 25% of all voice traffic use VoIP technologies, according to French regulator ARCEP”, commented Jacques Dunogué, SEVP of Thomson’s Systems Division. “Initiated to reduce telephony costs, it is now entering a second phase with services such as Fixed-Mobile Convergence. Thomson is proud to continue to be a technical leader in this area.”

“Thomson’s Cirpack softswitch and gateways have been on the market for almost 10 years, helping operators to expand their legacy PSTN infrastructure. The same Cirpack platform can also be used to deploy broadband telephony services with the features and regulatory requirements required for a primary-line voice service.” said Steve Byars of Current Analysis. “This not only allows operators to reduce OpEx and network complexity, it also lets them quickly launch VoIP services that can replace the PSTN. Today, the Cirpack softswitch is powering some of the largest European VoIP deployments and has demonstrated it represents an ideal solution for carriers’ multiple service-delivery needs while they migrate to next-generation networks and IMS." said Byars.

Source: voipcentral.org

There is no such service called Free in a strict sense. Even if you are making a free VoIP call, there are certain hidden costs associated with it, to say installation charge and Internet cost.

Any way, the availability of Skype-like applications have led us to classify free VoIP and paid VoIP. I dont need to define what free VoIP is.

A recent Ofcom report has candidly revealed that free VoIP services have failed attract a substantial number of customers in UK. I feel the scenario is same elsewhere.

The available of Skype like applications has no doubt drastically reduced the cost of international calls. However, they can not sideline the traditional phone service so easily for all practical reasons.

In its latest survey, the telecom regulatory authority has found that mere 17 per cent of adults in UK have used the services and only 14 per cent of people prefer to use the service over more conventional methods.

No doubt, the adoption of VoIP is growing at a rapid pace in the past few years. As per latest count, there are more than 2.4 million households across the UK having used VoIP service. It is only nine per cent of the total. Therefore, Ofcom says that the technology is still in the early-adopter phase.

Here I would like to raise a question that why the intake of VoIP is considerable low even if the adoption of broadband is growing rapidly.

There are number of reasons such as poor call quality, security, lack of general awareness, installment problem and lack of technical knowledge.

Source: voipcentral.org

VoIP has been changing the face of corporate communication than before. With IP technology, the business enterprises can create connecting links to their branch offices located in far distance places keeping touch with their workers. The cost of VoIP service is low, very low in contrast to the traditional phone services.

Thats why James Goodman, CEO of Digital Voice IP went to extend saying VoIP is an advance in telecommunications. It enables people to make and receive free calls over the internet, bypassing the call charges imposed by telephone companies.

With above assumptions, we tend to believe that VoIP is preferable to traditional Public Switched Telephone Network (PSTN). At the same time, we can not overlook security lapses of VoIP that mars the telecommunication service of a corporate sector. Security should be the priority for anyone contemplating enterprise VoIP deployments.

Security experts have traced new security threats of VoIP technology. They claim that software used for VoIP can be used by hackers to get into the data network and steal information. They have proved laptops running Windows XP SP2 with a Windows firewall and McAfee antivirus can be compromised enabling hackers to get through your network and delete or steal data off that laptop.
Rapid developments are taking place in the VoIP world with the launch of new products and services. However, it is half-done work. VoIP is still at its developmental stage. In the coming days, getting a secure VoIP network is a big thing for the business enterprises.

Source: voipcentral.org

In a country where 100,000 new users sign for a particular VoIP service (Read Skype) everyday, it is not surprising to announce sidelining of traditional phone services from mainstream. Yes, I am talking about China, which according to ReportLinker report, has enjoyed tremendous VoIP growth recording 109.931 billion minutes upto September 2006. It is 11.8 percent growth year over year.

Although, the call duration of VoIP in China is equal in number to that of PSTN, however the growth VoIP call duration is higher than PSTN. VoIP today accounts 43.16 percent in the long-distance call duration.

The research firm has claimed that the share of VoIP in the long-distance call market will be equal to or even exceed the total long-distance call of PSTN and mobile (GSM and CDMA) in the following two or three years gradually.

The compound annual growth rate of VoIP equipment spending by Chinese businesses will amount to 48% in the next five years. They will spend nearly 4.2 billion Chinese yuan renminbi in 2009.

Meanwhile, China also is suffering from illegal VoIP calls, which have been growing 30% every year from 2003 to 2006. Last year, illegal international call volume was about 500 million minutes. Therefore, the China government needs to take a cautious approach towards VoIP.

Image:

Source: snapvoip.blogspot.com

At First I thought it was Oma, meaning Grand Mother in Dutch! But it sounds the same if I am pronouncing it right.
So the Silicon Valley start up is coming out with VoIP service that offers a fancy gateway that is called a HUB and an extension called Ooma Scout. My Grand ma was a Girl Scout!
Anyway the Ooma has a unique approach. You pay in the form of purchasing the devices the HUB and the SCOUT and you get all local calls free. will also offer a free second line, conference calling, voice mail service and an online "lounge" where users may change their preferences or get voice mail in an e-mail format. More I investigate, hub is a Voip(broadband)/PSTN gateway with voice mail and some more bells and whistles added.
The Hub is said to be $399.00 and the company will start selling the devices Thursday with an invitation-only offer to select U.S. residents. I think it has to be by invitation because, a few will part with $400 after SunRocket took off with all those prepaid $199 a year accounts! and without recourse. It is not clear about the hub and the codecs and protocols it uses. Are the proprietary? Or are they compatible with any VoIP server like SER or Asterisk? If the answer is no to the latter, my $400 will stay in my pocket. I have too many VoIP gateways that I don’t use.(Can’t use!).

"It’s nothing like anything a carrier can do currently," CEO Andrew Frame said. "Once you own the box, you don’t have to pay ooma anything in the future." unless you make long distance calls.

Frame also says you have to pay again to Ooma, if you want to make long distance calls. the way they plan to make money. So far I know they offer 1 Cent per minute to Europe (Not clear if cellular calls included) and 8 cents to India. Not much different from other carriers.

So far I know very little about the technology and the methodology. But from guessing from the tid bits from various sources, one can achieve the same with Skype and some of the gateways listed at voip-info.org, consumer and enterprise versions are offered, they are much cheaper and could be used with any VoIP provider. Then there are a bunch of Skype Phones from the likes of Linksys and polycomm that cost less than $100 and plugs in to any broadband service!

The difference seem to be that Ooma will use your gateway (Hub) to terminate local calls to numbers that are not on Ooma network! We use to have such network in early 2000s when every one did not have broadband connections and wanted to make cheap VoIP calls. So we used to volunteer our phone line for greater good, until we found that software always did not find the correct termination number and called numbers beyond your free local call Zone, as pacbell used to call them! May be they have improved the technology! and a patent for improved technology?
Ooma seem to have patent-pending call-routing algorithm called "distributed termination," similar to peer-to-peer and distributed computing ideas. I would like to know how much is this different from that of Skype’s solution, which is also a distributed P2P other than hub having PSTN call capabilities.

Don’t take my word for it, follow this link to learn more about Ooma, (I could not, there is no much information) I would not call my users white rabbits, then again you can give a call to your Oma, Grandma, that you have been putting off, using any phone at hand!
I learned about it here.

Source: snapvoip.blogspot.com

At First I thought it was Oma, meaning Grand Mother in Dutch! But it sounds the same if I am pronouncing it right.
So the Silicon Valley start up is coming out with VoIP service that offers a fancy gateway that is called a HUB and an extension called Ooma Scout. My Grand ma was a Girl Scout!
Anyway the Ooma has a unique approach. You pay in the form of purchasing the devices the HUB and the SCOUT and you get all local calls free. will also offer a free second line, conference calling, voice mail service and an online "lounge" where users may change their preferences or get voice mail in an e-mail format. More I investigate, hub is a Voip(broadband)/PSTN gateway with voice mail and some more bells and whistles added.
The Hub is said to be $399.00 and the company will start selling the devices Thursday with an invitation-only offer to select U.S. residents. I think it has to be by invitation because, a few will part with $400 after SunRocket took off with all those prepaid $199 a year accounts! and without recourse. It is not clear about the hub and the codecs and protocols it uses. Are the proprietary? Or are they compatible with any VoIP server like SER or Asterisk? If the answer is no to the latter, my $400 will stay in my pocket. I have too many VoIP gateways that I don’t use.(Can’t use!).

"It’s nothing like anything a carrier can do currently," CEO Andrew Frame said. "Once you own the box, you don’t have to pay ooma anything in the future." unless you make long distance calls.

Frame also says you have to pay again to Ooma, if you want to make long distance calls. the way they plan to make money. So far I know they offer 1 Cent per minute to Europe (Not clear if cellular calls included) and 8 cents to India. Not much different from other carriers.

So far I know very little about the technology and the methodology. But from guessing from the tid bits from various sources, one can achieve the same with Skype and some of the gateways listed at voip-info.org, consumer and enterprise versions are offered, they are much cheaper and could be used with any VoIP provider. Then there are a bunch of Skype Phones from the likes of Linksys and polycomm that cost less than $100 and plugs in to any broadband service!

The difference seem to be that Ooma will use your gateway (Hub) to terminate local calls to numbers that are not on Ooma network! We use to have such network in early 2000s when every one did not have broadband connections and wanted to make cheap VoIP calls. So we used to volunteer our phone line for greater good, until we found that software always did not find the correct termination number and called numbers beyond your free local call Zone, as pacbell used to call them! May be they have improved the technology! and a patent for improved technology?
Ooma seem to have patent-pending call-routing algorithm called "distributed termination," similar to peer-to-peer and distributed computing ideas. I would like to know how much is this different from that of Skype’s solution, which is also a distributed P2P other than hub having PSTN call capabilities.

Don’t take my word for it, follow this link to learn more about Ooma, (I could not, there is no much information) I would not call my users white rabbits, then again you can give a call to your Oma, Grandma, that you have been putting off, using any phone at hand!
I learned about it here.

Source: snapvoip.blogspot.com

Test Results from over 14,000 test calls placed by Minacom’s PowerProbe 6000
to Western European and North American destinations from July 2005 to July 2006

Montreal, Canada (Monday, August 28th, 2006) — VoIP phone service now sounds better and connects faster than the standard public-switched phone network (PSTN), according to data collected over the last 12 months by Minacom’s standards-based, single-ended service quality test system. Results show that VoIP service quality increased steadily over the last year, with an average Mean Opinion Score (MOS) of 4.2, compared to 3.9 for the PSTN - MOS is a scale commonly used to describe speech quality, ranging from 1 (worst) to 5 (best). Based on a MOS threshold of 3.6, only 1 out of 50 calls in North America were considered to be unacceptable - 1 in 10 worldwide - while greater than 85% of VoIP calls exceeded average PSTN quality over the same period. Detailed results show that VoIP service bettered PSTN quality worldwide, and improved in all regions over the course of the survey. In addition to superior sound quality, calls over VoIP connected quicker overall - 8.2 seconds on average, compared to 8.9 seconds for those placed over the PSTN. Regionally, the PSTN was faster to connect for calls placed to North America (4.3 seconds vs. 5.7 for VoIP), while international calls connected faster with VoIP (8.7 vs. 10.4 seconds for PSTN). Linear regression indicates that VoIP is closing the gap, connecting 2 seconds faster in July 2006 than a year earlier.

A recent Internet Phone quality study by Brix Networks indicated that 1 in 5 calls were classified as unacceptable, and that call quality was steadily declining. As this study may have created the impression that VoIP service is not capable of delivering PSTN-grade phone service, Minacom felt it should be clarified for both those in the VoIP industry, and individuals and enterprises considering VoIP service, that this report evaluated computer-to-computer (PC-PC) Internet phone service, similar to those offered by Skype™, Google™ Talk, MSN™ and Yahoo™ Messenger. The quality and service reliability of these applications does not compare to that of the VoIP phone services offered by telcos, cable operators, and broadband VoIP providers who carefully deploy, monitor and manage the quality of their services. PC-PC VoIP quality is subject to many diverse impairments, including firewall settings, computer performance, antivirus installations, high-compression codecs, and Internet bandwidth shared with gaming, file downloads, web surfing and email. By contrast, VoIP offered by service providers is switched using telecom grade equipment, uses lower-compression codecs, and is prioritized over regular Internet traffic using sophisticated, standards-based multimedia telephone adapters (MTAs), maintained and monitored by the operator.

Minacom’s tests were conducted over PSTN, managed broadband and cable VoIP lines, the same services offered to residential and enterprise customers by phone, cable and hosted VoIP providers. Each month, Minacom’s PowerProbe® 6000 service level test probe places hundreds of calls from Minacom’s QoS labs in Montreal, Canada, to public destinations worldwide over PSTN, broadband VoIP, cable VoIP, DSL, FTTP and wireless networks, publishing the results in the Minacom QoS Benchmark Reports, a free email newsletter now in its fourth year of circulation. The results shown in this current study are based on the data published in these reports over a one year period from July 2005 to July 2006. Minacom’s QoS Benchmark Reports are used by the ITU Quality of Service Development Group in studies summarizing global phone quality, published annually to carriers worldwide for the consistency and accuracy of the measurements reported. Minacom’s Public Termination Inventory (PTI) database, used by the DirectQuality® R7 web-based test-OSS to automate the calls, contains over 200,000 far-end public numbers in 230 countries and administrative regions worldwide.

The human ear is an analog device, and sound is an analog signal, so it is important to include analog signal analysis when evaluating speech quality. Minacom’s DirectQuality R7 test system uses an award-winning combination of ITU and industry standard algorithms to calculate listening quality MOS using both analog and IP measurements. MOS scores based only on IP packet statistics do not capture the effects of echo cancellers in network equipment and telephone adapters, noise introduced by copper wiring, or issues with call volume and delay. Minacom’s PowerProbe 6000 IVR Test Agent measures a wide range of analog and IP impairments, including noise, echo, delay, packet loss, call volume, jitter and loss, as well as a complete array of connectivity metrics including Post Dial Delay (PDD), Answer Seizure Ratio (ASR), and Dial-Tone Delay (DTD). Minacom’s single-ended testing technology is used by multi-billion minute/year carriers worldwide to perform automated least-cost routing, validate partner carriers, monitor VoIP service quality and assure IP Peering SLAs.

“Carriers are becoming increasingly educated about MOS scoring and want to know where MOS scores are coming from.” commented Frost and Sullivan Telecom Industry Manager & Analyst, Jessy Cavazos, adding, “There are numerous products in the market that only look at the packet metrics. Hence, many carriers are starting to see degradation they should not see, or not seeing degradation they should see. False service quality alarms result in unproductive troubleshooting efforts by service providers, whereas unidentified quality issues ultimately leads to dissatisfied customers. That is why Minacom uses three different technology sources for MOS scoring instead of only one, so as to capture all possible service issues with the highest degree of accuracy available."
Press release