jump to navigation

Portsmouth, NH’s Whaleback Systems, a provider of managed VOIP services, raises $600,000 March 8, 2010

Posted by HubTechInsider in Mobile Software Applications, Startups, Telecommunications, Venture Capital, VoIP, VUI Voice User Interface.
Tags: , , , ,
add a comment

Portsmouth, NH’s Whaleback Systems, a provider of managed VOIP services, raises $600,000 from Ascent Ventures and Castile Ventures.

Add to FacebookAdd to DiggAdd to Del.icio.usAdd to StumbleuponAdd to RedditAdd to BlinklistAdd to TwitterAdd to TechnoratiAdd to Yahoo BuzzAdd to Newsvine

Add to Google Buzz

Get ready for high definition cellular and landline telephone calls November 3, 2009

Posted by HubTechInsider in Fiber Optics, Telecommunications, VUI Voice User Interface.
Tags: , , , , , , , , ,
add a comment
Uganda - mobile phone charging service

Image via Wikipedia

For over forty years, the quality of telephone calls has changed very little. The shift in the 1990’s from analog to digital cellular technology promised crisper quality, but the results never panned out. Struggling with 30% annual increases in cellular traffic, cellular telephone companies used the improved technology to add capacity, not improved quality.

Today the demand for cellular minutes is nearing its zenith, with mature growth levels of only 3% in the past year. Now the relentless advance of digital technology advances in cellular communications can be used for purposes other than simply packing more telephone calls into the cellular airwaves.

To this point in time, the big U.S. carriers plan to use their growing capability to provide all sorts of data services, but eventually, the cost of better sounding voice calls will be too cheap to ignore. Today’s carriers convert telephone calls into 6,000 digital bits per second, a tight squeeze and the major reason telephone calls sound so poor today. In the tiny European country of Moldova, French wireless carrier Orange has now deployed the world’s first high definition cellular telephone network, which uses double the number of bits per second. The highs and lows of the human voice are not so badly mangled using the high definition cellular telephone system.

In the U.S., chipmaker Broadcom is working on new equipment that will allow even better-sounding telephone calls. 32,000 digital bits per second will produce voice quality that is virtually indistinguishable from face-to-face conversation. The technology portends a clear audible improvement over not just ordinary cellular telephones but also landline telephones, which chop off high frequencies, especially above 3 kHz, the frequency range in which much human speech falls into.

Another big problem with cellular telephone calls is the annoying apparent lag that occurs between the moment when one caller speaks and the time his voice reaches the other person’s ear. Many people assume that’s an inherent drawback of cellular telephones, but it is not. Wireless digital cellular signals fly through the air at the speed of light just as they do in optical fiber – the delays come from slow software and circuitous routing. The new Long Term Evolution (LTE) gear set for deployment next year should cut that lag by at least 75%, so much that most human ears won’t notice it anymore.

Landline telephones stand to gain from the same quality advances as well. Orange has already installed 500,000 high definition landline telephones in Europe that use voice over internet technology (VOIP). When this style of telephone connection first hit the scene, it was roundly criticized for its poor sound quality relative to traditional landline telephones, but Orange and other carriers, some of whom are in the U.S. like Vonage, have shown that better technology can close that quality gap and then some. Both cellular telephone and Internet landline telephone calls may soon sound terrific as a result.

SS7 – Signaling System Seven – Telecommunications Protocol SS7 June 10, 2009

Posted by HubTechInsider in Fiber Optics, Telecommunications.
Tags: , , , , , , , ,
1 comment so far
SS7 software layers

SS7 software layers

In Signaling System 7 (SS7) protocol, a worldwide standard (with variations), routing intelligence is located in low cost computer-based equipment rather than in central office switches.

One of the primary benefits of SS7 is global interoperability. It has the capability to enable all carriers to cooperate with each other. It is a standard protocol approved by the ITU. Global billing, toll-free calling, 900-number services, and international wireless call roaming are all call features that are dependent on SS7.

SS7 is used on a global basis. In North America, the ANSI version of SS7 is used. In Europe, the ETSI version is used. In other pats of the world, the ITU version of SS7 is used.

Gateways allow these international SS7 implementations to communicate with each other.

SS7 is essential to modern networking. With SS7, an overlaid packet switched network controls the underlying voice network’s operation and signaling information is carried on a separate channel from voice and data traffic.

Because signaling is such a quick network activity, it is possible to multiplex many signaling messages over one signaling channel using a packet switching arrangement.

SS7 permits the telephone company to provide one database for several switches in order to freeup switch capability for other functions. This is the capability that makes SS7 the foundation for Intelligent Networks (INs) as well as Advanced Intelligent Networks (AINs).

As an example, in order to provide a service such as 900 number and toll-free calling, in SS7, powerful parallel processing computer systems hold massive databases with information such as routing instructions for toll-free and 900 number telephone calls. One processor with its database supports many central office switches under SS7. in this way, each central office itself is not required to host the centralized database. Without the need to share the expense of maintaining the sophisticated routing information, each central office can share in the expense of a database or feature upgrade to the centralized SS7 datastore.

MCI first implemented SS7 into its network in 1988. SS7 enabled them to halve their call setup time on calls between Philadelphia and Los Angeles. Freeing up voice channels from their previous signaling duties pre-SS7 enabled carriers to pack more voice calls on their existing network paths.

Cellular networks use SS7 technology to support roaming. Every cellular provider has a database called the home location register, or HLR, where complete information regarding each subscriber is kept. They also maintain a database called the VLR, or visitor location register, that maintains information on each caller who visits from other areas. When a cellular subscriber roams, each network they visit exchanges SS7 messages with their “home” network. The subscriber’s home system also marks its HLR so that it knows where to send calls for its customers who are roaming.

SS7 has three major components:

1. Packet switches – Signal Transfer Points that route signals between databases and central switches. STPs, or Signal Transfer Points, are responsible for translating the SS7 messages and then routing these messages amongst the various network nodes and databases. Signal Transfer points are packet switches that route signals between central offices as specialized databases. Messages are sent between points on the SS7 network in variable-length packets with the addresses attached. Signal transfer switches read only the address portion of the packets and forward the messages accordingly.

2. Service Switching Points – Software and ports in central offices that enable switches to query databases. SSPs are the switches that begin and end calls. They receive signals from the Customer Provided Equipment (CPE) and then process the calls on the behalf of the end users. The user triggers the network to provide various services by dialing particular digits. SSPs are typically implemented at access tandem offices, local exchanges or toll centers that contain the needed network signaling protocols. The SSP serves as the begining and ending point for SS7 messaging.

3. Service Control Points – DBs with customer feature and billing information. Service Control Points, or SCPs, interface with SSPs as well as STPs. The STP contains the network configuration and call-completion database – the SCP contains all the service logic that is needed to deliver the type of call and feature in the call that the user is requesting. SCPs are centralized network nodes that contain software and databases needed for call management. Functions such as digit translation, call routing and verification of credit cards are all provided by SCPs. Usually a SCP will receive traffic from a SSP via the STP and will then return responses based on those queries by way of the STP.

The SS7 signaling data link is a full duplex digital transmission channel that operates at either 56 Kbs (T-Carrier transmission systems, in North America) or 64 Kbps (E_Carrier transmission systems, Europe). SS7 also defines a number of other types of links, each with a specific use within a SS7 network.

A (access) links
B (bridge) links, D (diagonal) links, and B/D links
C (cross) links
E (extended) links
F (fully associated) links

Geek T-Shirts, Decals, and more at http://www.tshirtnow.net

Want to know more?

You’re reading Boston’s Hub Tech Insider, a blog stuffed with years of articles about Boston technology startups and venture capital-backed companies, software development, Agile project management, managing software teams, designing web-based business applications, running successful software development projects, ecommerce and telecommunications.

About the author.

I’m Paul Seibert, Editor of Boston’s Hub Tech Insider, a Boston focused technology blog. I have been working in the software engineering and ecommerce industries for over fifteen years. My interests include computers, electronics, robotics and programmable microcontrollers, and I am an avid outdoorsman and guitar player. You can connect with me on LinkedIn, follow me on Twitter, follow me on Quora, even friend me on Facebook if you’re cool. I own and am trying to sell a dual-zoned, residential & commercial Office Building in Natick, MA. I have a background in entrepreneurship, ecommerce, telecommunications and software development, I’m a Technical PMO Director, I’m a serial entrepreneur and the co-founder of several ecommerce and web-based software startups, the latest of which are Twitterminers.com and Tshirtnow.net.

Add to FacebookAdd to DiggAdd to Del.icio.usAdd to StumbleuponAdd to RedditAdd to BlinklistAdd to TwitterAdd to TechnoratiAdd to Yahoo BuzzAdd to Newsvine

What is the Mu-Law PCM voice coding standard used in North American T-Carrier telecommunications transmission systems? June 8, 2009

Posted by HubTechInsider in Definitions, Telecommunications, VUI Voice User Interface.
Tags: , , , , , , , , ,
1 comment so far
Sampling and 4-bit quantization of an analog s...

Image via Wikipedia

Mu-Law encoding is the PCM voice coding standard used in Japan and North America. It is a companding standard, both compressing the input and expanding the data upon opening after transmission. Mu Law is a PCM (Pulse Code Modulation) encoding algorithm where the analog voice signal is sampled eight thousand times per second, with each sample being represented by eight bits, thus yielding a raw transmission rate of 64 Kps. Each sample consists of a sign bit, a three bit segment which specifies a logarithmic rqange, and a four bit step offset into the range. The bits of the sample are inverted before transmission. A Law encoding is the voice coding standard which is used in Europe.

Want to know more?

You’re reading Boston’s Hub Tech Insider, a blog stuffed with years of articles about Boston technology startups and venture capital-backed companies, software development, Agile project management, managing software teams, designing web-based business applications, running successful software development projects, ecommerce and telecommunications.

About the author.

I’m Paul Seibert, Editor of Boston’s Hub Tech Insider, a Boston focused technology blog. You can connect with me on LinkedIn, follow me on Twitter, even friend me on Facebook if you’re cool. I own and am trying to sell a dual-zoned, residential & commercial Office Building in Natick, MA. I have a background in entrepreneurship, ecommerce, telecommunications and software development, I’m the Senior Technical Project Manager at eSpendWise, I’m a serial entrepreneur and the co-founder of Tshirtnow.net.

What is the frequency response of the North American Public Switched Telephone Network? June 3, 2009

Posted by HubTechInsider in Telecommunications, VoIP, VUI Voice User Interface, Wireless Applications.
Tags: , , , , , , ,
1 comment so far
Sinusoidal waves of various frequencies; the b...

Image via Wikipedia

The conventional North American Public Switched Telephone Network, or PSTN, has a frequency response range of 300 Hz to 3,400 Hz. The normal hearing range of humans is typically 30 Hz to 20,000 Hz. So the conventional telephone transmission system is unable to carry bright, high-frequency and deep, low-frequency tones.

But, somewhat surprisingly, because our ears are so used to hearing poor-quality audio over the telephone, our brains actually “fill in” the missing frequencies. As an example, the crisp “s” sound in the word “Christmas”. So in effect, the telephone audio often sounds better than it actually is to us.

An explanation of the Nyquist Theorem and its importance to Mu-Law Encoding in North American T-Carrier Telecommunications Systems June 2, 2009

Posted by HubTechInsider in Definitions, Fiber Optics, Mobile Software Applications, Telecommunications, VUI Voice User Interface, Wireless Applications.
Tags: , , ,
add a comment

nyquist100

The Nyquist theorem established the principle of sampling continuous signals to convert them to digital signals. In communications theory, the Nyquist theorem is a formula stating that two samples per cycle is all that is needed to properly represent an analog signal digitally. The theorem simply states that the sampling rate must be double the highest frequency of the signal. So, for example, a 4KHz analog voice channel must be sampled 8000 times per second. The Nyquist Theorem is the mathematical underpinning of the Mu-Law encoding technique used in T-Carrier transmission systems. T-Carrier is used in North American telecommunications networks. In Europe, where E-carrier transmission systems are used, a similar but incompatible theorem, Shannon’s Law, is used in the European A-Law encoding technique. This is the reason why Mu-Law encoding is used in North America and A-Law encoding is used in Europe.

The author of the Nyquist Theorem was named Harry Nyquist. Harry worked in the research department at AT&T and later at Bell Telephone Laboratories. In 1924, he published a paper titled “Certain Factors Affecting Telegraph Speed”, which analyzed the correlation between the speed of the telegraph system and the number of signal values it used. Harry refined his paper in 1928, when he republished his work under the title “Certain Topics in Telegraph Transmission Theory”. It was in this paper that Harry expressed the Nyquist Theorem, which established the principle of using sampling to convert a continuous analog signal into a digital signal. Claude Shannon, the author of Shannon’s Law, cited both of Nyquist’s papers in the first paragraph of his classic paper “The Mathematical Theory of Communication”. Harry Nyquist is also known for his explanation of thermal noise, also sometimes known as “Nyquist noise” as well as AT&T’s 1924 version of a fax machine, called “telephotography”.

His remarkable career included advances in the improvement of long-distance telephone circuits, picture transmission systems, and television. Dr. Nyquist’s professional, technical, and scientific accomplishments are recognized worldwide. It has been claimed that Dr. Nyquist and Dr. Claude Shannon are responsible for virtually all the theoretical advances in modern telecommunications. He was credited with nearly 150 patents during his 37-year career. His accomplishments underscore the excellent preparation in engineering that he received at the University of North Dakota. In addition to Nyquist’s theoretical work, he was a prolific inventor and is credited with 138 patents relating to telecommunications.





Want to know more?

You’re reading Boston’s Hub Tech Insider, a blog stuffed with years of articles about Boston technology startups and venture capital-backed companies, software development, Agile project management, managing software teams, designing web-based business applications, running successful software development projects, ecommerce and telecommunications.


About the author.

I’m Paul Seibert, Editor of Boston’s Hub Tech Insider, a Boston focused technology blog. You can connect with me on LinkedIn, follow me on Twitter, even friend me on Facebook if you’re cool. I own and am trying to sell a dual-zoned, residential & commercial Office Building in Natick, MA. I have a background in entrepreneurship, ecommerce, telecommunications and software development, I’m the Senior Technical Project Manager at eSpendWise, I’m a serial entrepreneur and the co-founder of Tshirtnow.net.

Add to FacebookAdd to DiggAdd to Del.icio.usAdd to StumbleuponAdd to RedditAdd to BlinklistAdd to TwitterAdd to TechnoratiAdd to Yahoo BuzzAdd to Newsvine

Why designing for a VUI is more difficult than designing for a GUI May 11, 2009

Posted by HubTechInsider in Mobile Software Applications, VoIP, VUI Voice User Interface, Wireless Applications.
Tags: , , , , , ,
4 comments

Despite the fact that many Automated telephony and IVR vendors advertise that their web-based SaaS offerings can seemingly make the development, testing, deployment and maintenance of an IVR application seem easy and straightforward, this over-confidence in the VUI design abilities of untrained, non-technical business analysts and enterprise services managers is woefully misplaced. This mistaken impression is borne out by the simple fact that just because a software tool may be easy to use (even though all of these SaaS web-based vendors provide VUI tools with horrific interfaces and GUI designs, such as reliance on stone-age Java applets) only cursory thought, if any thinking at all, has been invested into how these untrained resources should use that tool. This can and often does lead to catastrophic results.

I frequently encounter the mistaken prevailing notion that designing a VUI consists of nothing more than taking a GUI and “simplifying it” for use on the telephone. As the thinking goes, we can all talk on the telephone; Not all of us can navigate a complex forms-based web site. But despite this mistaken general impression (perpetuated by IVR and automated telephony vendors and many software development teams within them, as well as their clients), some basic realities persist in shattering these ill-conceived concepts: People can read faster than they can listen with comprehension, speak faster than they can type, and talk much more quickly than they can process the meaning behind spoken words. So even though, based on initial impressions, designing an effective VUI might seem easier than designing a first-rate GUI, the converse is true: designing a great VUI is far more difficult than designing a GUI.

A VUI is inextricably linked with Time

When a user is navigating a GUI, they can read text at any location on the web page or application screen. The user can skip ahead visually to the section they are interested in. With a VUI, the user is a “prisoner” of the VUI design. The attention is captive: they must listen with (or without) patience to each word before they can hear the one that follows it. With this in mind, some best practices for VUI design emerge:

1. Long prompts are Bad: The longer the prompt, the more the user’s patience is being taxed. Introductory or “tutorial” prompts explaining how the system works may be required for an outbound IVR application or alternatively provided for the benefit of novice users, however they should not be forced upon returning visitors or outbound IVR call recipients that have received similar IVR communications in the past.

2. Long VUI menus are Bad: Again to use the GUI as a contrasting example, on a web page you can present many menu options to the user, even hiding numerous options in a drop-down menu. A VUI menu, on the other hand, should never exceed five or six items at the most.

3. Get to the gist of the communication quickly: Forcing your captive “audience” to listen through introductory marketing copy written into an outbound IVR or inbound VRU script will become annoying very quickly to the user. Script your important information into the beginning of your prompts.

4. Allow ‘barge-in’: Expert users who know how to use the system and know what they want to do desire the ability to speed up the automated interaction with the system. Allow them to issue their commands to the system without forcing them to wait for the system to finish talking.

5. Give expert users global hotwords: Global “hotwords”, or application-level shortcuts, allow users to “cut to the chase”, enabling them to cut through menus and enjoy the feeling of enablement that a responsive VUI system can provide.

6. Allow the user to pause the interaction: The GUI has another crucial advantage over the VUI – the ability to stop and start again exactly where you left off after an indeterminate interval. While providing the exact same level of interaction control to the user is impossible in a VUI, if within your VUI design you are asking the user to provide the system with a membership number in a COB (Coordination of Benefits) automated telephony call for a health care provider, or asking them for their account number in an inbound VRU application, or if the system wants the user to write down a confirmation code or other information, then design your VUI so that the call recipient or caller can get their pencil and paper ready, find their membership card, and say “continue” when they are ready.

The One-way Temporal Flow of the User

Of course, the spoken word is not only temporally linear, but also one-way. In the same manner in which time is a “one way street”, so is speech a “one way medium”. When you are listening to a prerecorded voice prompt, you can’t easily hit the nonexistant rewind button on your telephone. A VUI is not like watching a ball game on your DVR or Tivo, either. You can’t easily go back and listen to the prompt again. This is in stark contrast to the GUI world, where the user can jump back-and-forth within the text on the page or screen. Three simple techniques can help to alleviate this conundrum:

1. Always let the user ask to have the system repeat the prompt: Perhaps the most elementary technique to mitigate the one-way temporal flow of the user is to have the system offer to repeat the last prompt. The user must be made aware of the fact that they can have any prompt repeated to them at any time during the IVR interaction.

2. Make Help available to the user: Information or instructions that are crucial to the task completion ability of the call recipient or caller presented at the beginning of the interaction must be made available to the user at any point in the IVR interaction. Offer help to the user not only at the beginning of the call but also at moments where the user seems to have arrived at an impasse in the interaction. The need to offer help to the user is acute at “no input”, “Out of Grammar (OOG)” or “no match” states.

3. Present a summation of the gathered data: In form-filling dialogs or IVR interactions where the caller is being asked to provide information to the system, a marvelous approach to overcome the one-way temporal flow nature of the IVR interaction is to offer the call recipient or caller a summation of the data that has been gathered from them during the course of the IVR interaction so far.

Persistence in a VUI is not visible to the user as in a GUI

Callers or call recipients perhaps show the most frustration when they feel they have lost track of “where they are” in the course of traversing a scripted IVR inbound or outbound interaction. Aggravation mounts as the user becomes increasingly unsure of what to do next, and what the system expects the user to do next. Whereas a web page or application screen typically provides a multitude of visual ques, such as a menu tree, “breadcrumb” navigation path, or something similar, even something as simple and effective as a URL web address window on a browser is unavailable in the VUI world. Some approaches to mitigate these factors emerge to the experienced VUI designer:

1. Auditorily “Announce” the user’s position in the IVR exchange: In the same manner that a properly designed web page or application screen will tell the caller or call recipient where they are in terms of navigating a site or application, so should a well-designed voice interface let the user know their exact position in the IVR interaction. A simple and efective technique for providing the user with such “mental markers” is to use a word or two to announce this position to the user: “Main Menu”…”Here’s the drugs in your prescription refill:”, etc.

2. Audio breadcrumbs: The VUI version of the “breadcrumb navigation” trails that are featured so prominently on web sites in the GUI world can be emulated in the VUI world, where they prove no less useful. Each “voice page” that requires interaction with the user can be associated with a “position page” that announces the user’s position within the dialog tree. “Prescription, Reorder, Address”, as an example, would very nicely indicate to the user that they chose “prescriptions”, then “Reorder”,a nd are now confirming their prescription reorder address on file with the system. A “Go Back” provision or option should be offered to users at these “position page” states.

3. Audio Icons: Auditory icons, or “earcons”, are VUI equivalents of the GUI’s icons. These audio icons can be extremely useful to both the VUI designer as well as the call recipient or caller by either annoucing to the user that a particular action is about to be undertaken or positioning the user within a IVR menu structure or transaction path. “Wait audio”, or sounds played to the user to indicate that the system is busy performing a record lookup or other function can prevent the user from interpreting a system crash or IVR interaction end when faced with an absolute extended silence.

GUIs present one fundamental advantage over VUIs: the user navigating a web page or an application screen has control over the medium, the message, and the interaction itself. Although a poor GUI can make the user feel helplessly confused, a VUI faced with the challenges outlined above has to be near-perfect to prevent the user abandoning the IVR interaction entirely by the simple and universal act of hanging up the telephone. VUI designers should always be aware of the significant differences between designing an effective and useful GUI and VUI. It would be ill-advised to enter into a VUI design task or project of any size while carrying into the endeavor the familiar GUI design assumptions.

Want to know more?

You’re reading Boston’s Hub Tech Insider, a blog stuffed with years of articles about Boston technology startups and venture capital-backed companies,software developmentAgile project managementmanaging software teams, designing web-based business applications, running successful software development projectsecommerce and telecommunications.

About the author.

I’m Paul Seibert, Editor of Boston’s Hub Tech Insider, a Boston focused technology blog. You can connect with me on LinkedIn, follow me on Twitter, even friend me on Facebook if you’re cool. I own and am trying to sell a dual-zoned, residential & commercial Office Building in Natick, MA. I have a background in entrepreneurshipecommercetelecommunications andsoftware development, I’m the Director, Technical Projects at eSpendWise, I’m a serial entrepreneur and the co-founder of Tshirtnow.net.

Add to FacebookAdd to DiggAdd to Del.icio.usAdd to StumbleuponAdd to RedditAdd to BlinklistAdd to TwitterAdd to TechnoratiAdd to Yahoo BuzzAdd to Newsvine

Why IP beat out ATM for use in Next-Generation Voice Networks May 10, 2009

Posted by HubTechInsider in Telecommunications.
Tags: , , ,
add a comment

For a good part of the 90’s, conventional wisdom in the telecommunications industry held that asynchronous transfer mode (ATM) and Internet Protocol (IP) were competing technologies. IP, the prevailing notion held, was a “best effort” service because IP-based networks indiscriminately discarded packets if there was congestion. There was no standardized protocol to identify and prioritize video and voice. The industry at that time maintained that best effort protocols would not be recognized by carriers as acceptable for voice traffic. Because of this, ATM’s ability to create virtual connections and to prioritize voice and video so that packets would never be dropped and quality of service standards were met gave ATM a vital advantage.

ATM also had speed advantages, capable of speeds of 155 and 622 megabits per second. Ethernet LANS at this time were limited to 10 megabits per second, and IP used between networks was also slower than ATM. For these above reasons, when carriers wanted to improve their networks, they decided on ATM equipment. I personally was involved in Fleet Bank’s multi-million dollar loan to LDDS Worldcom (now MCI) in the late nineties for ATM gear for their UUNET data network subsidiary.

However, despite all of its inherent advantages, ATM gear was costly and complex to install. There was a slight push around this time for ATM to be used in LANs, especially in campus backbone networks and NSF research nets, but ATM was far too expensive to deploy on the desktop. So ATM was relegated to use in large corporate backbone networks and carrier traffic-bearing data networking.

So, as you can imagine, mainly due to the speed and quality-of-service advantages, established telecom vendors and most new softswitch vendors initally at least based their next-generation voice switch architecture on ATM rather than IP. Meanwhile, improvements in routers and faster speeds on IP networks were making IP networks much more suitable for voice. Also at this time, Cisco’s TAG protocol, the forerunner of today’s MPLS, was being developed and was maturing. The MPLS protocol marked packets so that voice and video could be prioritized. This capability let IP packet flows be handled similarly to ATM virtual connections, which treat various types of traffic differently. Concurently, IP speeds improved from 10 megabits per second to 100 megabits per second speeds and, eventually, gigabit speeds.

With these notable improvements in speed and service qualities, along with the fact that corporate endpoints were already equipped to deal with IP traffic, the founders of Sonus Networks (Westford, MA), in 1997, choose to base their next-generation, softswitch-based voive infrastructure on IP. In this manner, Sonus was granted a head start over competitors who initially developed platforms based on ATM, losing time and previously invested development money when they switched over to IP – too late.

In related news, Sonus Networks of Westford, MA recently (11 March 09) announced it is “restructuring” again, cutting another 60 employees to complete its third round of cuts in three months. The company said this cut will equal out to about 6% of their workforce. The total job cuts within the three months has added up to 160 jobs lost at the networking equipment vendor. Sonus has a baseline resource level of approximately 1,000 people.

BT (British Telecom) has also recently (3 May 09) announced that it is cutting back on deployments of equipment and resources for its 21CN Next Generation Network (NGN) project.

Jefferies & Company analyst George Notter points out in a recent research note that Sonus was slated to in late 2007 to provide an Access Gateway Controller Function (AGCF) to enable communications between core IP and PSTN access networks. However, Sonus may have to take a revenue hit now, as BT discovered that its NGN network architecture is too costly. They have halted the NGN Network cutover project.

Update: Sonus Networks announces 2009 Q1 results

How telephone numbers are assigned May 3, 2009

Posted by HubTechInsider in Definitions, Telecommunications.
Tags: ,
add a comment

cellphones

The North American Numbering Plan Administration assigns telephone numbers to state-certified wireline carriers in each state. Wireless carriers also receive numbers from the North American Number Plan Administration. However, they don’t need to register on a state-by-state basis because the FCC, not individual states, licenses them to offer service. Carriers such as Vonage, Broadview Networks, and SBC for their IP services are required to obtain telephone numbers from local exchange carriers (LECs) in each state. The LECs can be either the incumbent or a competitor to the incumbent. The reason for this requirement is that VoIP is not defined at this time as a telecommunications service. Thus, VoIP carriers or the department and subsidiaries within carriers that offer VoIP must enter into agreements with a licensed carrier to obtain local telephone numbers in each state in which they wish to offer Voice over IP service. SBC IP has asked the FCC for a waiver of the requirement to obtain numbers from other carriers. In their own territory, they receive num,bers from their parent, SBC. However, when they offer VoIP outside of their home territory, they have to enter agreements with other LECs. Prior to the announced merger with SBC, AT&T objected to SBC IP’s request for a waiver, saying this would be unfair to other VoIP providers.


The North American Numbering Plan Administration assigns numbers in blocks of 1,000. This is called the number pooling system of allotting numbers because pools of 1,000 unused numbers are created. Prior to the year 2000, numbers were assigned to carriers in blocks of 10,000. This resulted in wasted numbers because many smaller carriers who did not use up all of their numbers could not share them with other carriers. To further conserve their numbers, in 2000, the FCC mandated that phone companies must first use up 60% of their assigned phone numbers before being given new ones. As of June 30, 2004, that percentage increased to 75%.

Measuring Voice Quality in a VoIP environment May 1, 2009

Posted by HubTechInsider in Telecommunications, Uncategorized.
Tags: , ,
add a comment

One of the consequences of installing Voice over IP systems is that the “voice” sides of information technology departments are learning the lingo and technology of measuring voice quality on data networks. In addition, staffs that manage data networks are becoming aware of the criticality of voice. They are developing a cognizance of the impact on voice services of congestion when they add new applications. They also note lost voice service when they take down the network for maintenance or new installations.

Staff use network management tools that entail quality of service assesments to monitor the following factors in voice quality:

* Packet loss refers to the network dropping packets when there is congestion. Packet loss results in uneven voice quality. Voice conversations “break up” when packet loss is too high.

* Latency refers to delays when voice packets transverse the network. Latency is measured in milliseconds. It results in long pauses within conversations and clipped words.

* Jitter is uneven latency and packet loss resulting in noisy calls that contain pops and clicks or crackling sounds.

* Echo, hearing your voice repeated, is often caused when voice is translated from a circuit switched format to the IP format. This is usually corrected by special echo-canceling devices.

The differences between IPT, Internet Telephony, and VoIP April 22, 2009

Posted by HubTechInsider in Definitions, Telecommunications.
Tags: , ,
add a comment

People use several IP-realted terms interchangeably. However, according to the International Telecommunications Union (ITU; http://www.itu.int), there are distinctions between the following terms:

* IPT – The transmission of voice, fax, and related services over a packet-switched IP-based network. Internet telephony and VoIP are specific subsets of IPT.

* Internet Telephony – Telephony in which the principal transmission network is the public internet. Internet telephony is commonly referred to as Voice over the Net, Internet phone, and net telephony, with appropriate modifications to refer to fax as well, such as Internet Fax.

* VoIP – IPT in which the principal transmission network or networks are private, managed IP-based networks.





Want to know more?

You’re reading Boston’s Hub Tech Insider, a blog stuffed with years of articles about Boston technology startups and venture capital-backed companies, software development, Agile project management, managing software teams, designing web-based business applications, running successful software development projects, ecommerce and telecommunications.


About the author.

I’m Paul Seibert, Editor of Boston’s Hub Tech Insider, a Boston focused technology blog. You can connect with me on LinkedIn, follow me on Twitter, even friend me on Facebook if you’re cool. I own and am trying to sell a dual-zoned, residential & commercial Office Building in Natick, MA. I have a background in entrepreneurship, ecommerce, telecommunications and software development, I’m the Senior Technical Project Manager at eSpendWise, I’m a serial entrepreneur and the co-founder of Tshirtnow.net.

%d bloggers like this: