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internet telephony


Twenty-four million pay for VoIP

Tuesday, January 09, 2007

Twenty-four million pay for VoIP

Twenty-four million people worldwide pay to use retail voice over IP (VoIP) services, new research has shown.

The number of retail VoIP subscribers jumped more than 80 per cent to 18.7 million during 2005 - or 24 million if you add users of PC-to-phone services such as Skype, according to Point Topic's IP Telephony report.

Based on reported revenues, the analyst estimates that one-sixth of VoIP subscribers used Skype in the fourth quarter of 2005.

The Asia-Pacific region has the largest share of VoIP users but the fastest growth is seen in Europe and the US.

The number of VoIP subscribers in Europe grew nearly threefold to 5.3 million during 2005 and is expected to pick up even more in 2006.

France has the highest VoIP adoption in Europe with 2.8 million paying subscribers, led by new carriers Free and neuf selling easy-to-use VoIP services - though incumbent France Telecom's VoIP push has also proved fruitful as it raised subscriber levels more than fivefold in 2005.

In the UK, BT's VoIP offering had "relatively few" subscribers by the end of 2005 but was picking up in early 2006, said the analyst.

One key success in the US, according to Point Topic, is marketing the service not as 'VoIP' but describing it in simple terms customers understand such as 'digital telephone'.

RIM Mum On Stock Options Review

Monday, January 08, 2007

RIM Mum On Stock Options Review

The NY Times had a story on Christmas day regarding a RIM stock options review. RIM, maker of the BlackBerry, initiated the review in Sep 2006 themselves but have not provided investors or regulators much in the way of details. Even the letter submitted to the OSC (province of Ontario Securities Commission) didn't clear things up much. The stock jumped approximately US$50 between Sep and Dec. More details at the NY Times (free registration may be required).

RIM got out of a patent infringement lawsuit earlier this year and recenty levied a copyright infringement lawsuit on Samsung's Blackjack device, amidst a great Q3 2006 in terms of revenues. Regardless of the hubbub, RIM seems like teflon, able to weather the goings on, fair or otherwise. I've met former employees who retired early because their stock options made them young millionaires. My feeling is that the company will continue to make millionaires for a long time because of good overall management and vision. [I own no RIM stock, but may purchase some in late 2007.]


Video Killed The Mobile Phone?

ABI Research released a report in late December stating that the mobile video market in mainland China will pass 32Mln users in 2008. The significance of this is tied to the fact that the Beijing Olympics takes place in that year. There will be two different technologies in use: broadcasting (27%) and unicast streaming (73%). Some users are expected to use both technologies. The Chinese SARFT (State Administration of Radio, Film, and Television) announced two voluntary standards last year: DAB, which will give way to T-DMB. More details at ABI Research.

I'm still the Doubting Thomas on video about mobiles. Is there enough bandwidth? Are screens even big enough to prevent eyestrain, and to thus be enjoyable? While mobile video use will undoubtedly increase in the next few years, there's a difference between conducting short video calls and watching prolonged mobile video. But if there's any candidate for enjoyable mobile video, I'd have to say it's likely the Nokia N-series of phones - though I haven't laid my mitts on one yet.


Communication Breakdown: 5 Fake New Year's VoIP Resolutions

VoIP/ IP comm and related companies have made some boo-boos this year, and are probably making some resolutions for the New Year. Here's what they might be resolving to do.

  1. Skype. We resolve to put more thought into our business version's design, and actually let bloggers know about new versions ahead of time. Well, at least a day or two.
  2. Vonage. We resolve to boost our stock's share price to at least $15 in 2007.
  3. Gizmo Project. We resolve to stop being the Rodney Dangerfield of VoIP services and get some respect. And more subscribers.
  4. Google. We resolve to think through our click-to-call products before releasing them.
  5. Cable companies. We resolve to stop screwing subscribers with ridiculous residential VoIP rates when we're already screwing them on TV subscription rates.

VOIP Information

Wednesday, January 03, 2007

VOIP Information

Voice Over IP ? Saving Money

I was on a tech support call with a client in Australia for over forty-five minutes. Guess how much that cost me. If I told you less than a dollar, would you believe it? Well it is true indeed.

VoIP - A Laymans Look - Should You Or Shouldnt You?

"As business people we manufacture shin pads, or we distribute cat food, or we evangelize, but should we be considering VoIP? Will it make us more money, or save us time?" -Dennis Schooley


This article contains the excellent information about the Voice Over IP Protocol.

Use your Computer to make Cheap Long-Distance Telephone Calls

What is ?VoIP??

Voice over IP Offers Your Business an Inexpensive Alternative To Toll Free Numbers

New businesses can now let their customers contact them for free or inexpensively with the use of Voice over IP telephony, without having to lease the ever more expensive toll free numbers offered by many telecoms companies around the world. With Voice over IP telephony, local numbers can be setup in scores of towns and cities across a country at relatively low costs. These can then connect to a central Voice over IP service provider, which will allow free or low rate calls to you from your customers.

Will VoIP be a Mass Market Product?

A common thinking among "Marketing people " is that for every product that enters the market there must be a path, a target, a need ( real or created) that decides how the product must enter the consumer's life, which part of the population is more likely to go for it, which niche it is going to fill and, most important "...certain things being stated, something other than what is stated follows of necessity from their being so." and that is the final issue: the price.

World On IP Community versus Telecoms Monopoly

World on IP community versus the TELECOMS' monopoly or a dream of a visionary

How Voip Can Mask The Size Of Your Business And Save You A Bundle

Have you heard the buzz about VOIP (Voice Over Internt Protocol)? Basically, it's like communicating over the phone without a phone. Instead you plug your microphone into your computer log on to a website and "boom" you're talking baby. You may be saying "Hasn't this been around for a while?", and you'd be right. But with the advances recently made to internet technology this once "nice to have" product available to everyone, dial-up or broadband. No long distance charges, no entry fees, no codes to remember and even better, with the better services, you can share applications and web browsers.

Making PC to Phone VoIP Calls over Dial-Up Internet Connections

There?s a lot of hype these days surrounding Internet-based voice communications (VoIP) replacing traditional telephone service. Most of this revolves around companies such as Vonage who coin themselves as the ?Broadband Phone Company?. So what about those of us who don?t have a broadband connection? Just because you don?t have high speed Internet, doesn?t mean that you can?t save a fortune by using VoIP for your long-distance calls. Whilst a broadband connection will usually result in more consistent VoIP call quality, comparable results can be achieved using a dial-up connection provided some simple guidelines are followed.

Do You Text Chat (IM) Online?

IM (Instant Messaging) - man, we thought we had it made in the shade when we stumbled over this unique way to communicate! Outside of talking person-to-person on the phone, IM was the next best thing. Texting talk was endless and cost was nonexistent. Well, those were the good old days of yore ? you know, those days of glorious innocence.

Voip Products

Tuesday, January 02, 2007

Customer Contact


Optical Networks

Phones, Clients & Accessories

Security & VPN

VoIP & Multimedia Communications

Wireless Networks

Bandwidth Consideration

Bandwidth Consideration

From all we said before we noticed that we still have not solved problems about bandwidth, how to create a real time streaming of data.

We know we couldn't find a solution unless we enable a right real-time manager protocol in each router we cross, so what do we can do?

First we try to use a very (as more as possible) high rate compression algorithms (like LPC10 which only consumes a 2.5 kbps bandwidth, about 313 bytes/s).

Then we starts classify our packets, in TOS field, with the most high priority level, so every router help us having urgently.

Important: all that is not sufficient to guarantee our conversation would always be ok, but without an great infrastructure managing shaping, bandwidth reservation and so on, it is not possible to do it, TCP/IP is not a real time protocol.

A possible solution could be starts with little WAN at guaranteed bandwidth and get larger step by step.

We finally have to notice a thing: also the so called guaranteed services like PSTN line could not manage all clients they have: for example a GSM call is not able to manage more that some hundred or some thousand of clients.

Anyway for a starting service, limited to few users, VoIP can be a valid alternative to classic PSTN service.

VoIP Communication Using PSTN line

VoIP Communication Using PSTN line


VoIP becomes very interesting when you start to use PSTN lines to call other people in the world, directly to their home telephone.


A typical application is like that:

Home phone1 - (PSTN) - PC1 - (Internet) - PC2 - (PSTN) - Home phone2

  1. Home Telephone1 make a calls to PC1 phone number (using PSTN line, I mean classic telephone line).
  2. PC1 answer to it.
  3. Home telephone1 must tell PC1 what gateway use (PC2 in this case) by giving the IP address (from DTMF keyboard) and/or what number call (in this case Home telephone2).
  4. After that PC1 will start to make an H323 call to PC2, then it will pass Home telephone2 to PC2 to make it call it throught PSTN line.
  5. Home telephone2 answers to call and communication between Home telephone1 and Home telephone2 begins.

What can be changed in this configuration?

  1. You may use a PBX to select many lines to access many PC1 gateway (for example one to call within your state, one to go accross Europe, and so on...): typically you don't have to change this, cause cost is always the same.
  2. You can select (after called your PC1 gateway) every PC2 you want (for example a PC2 living in England to call an English person so that you'd pay only intra-country costs).

So your decision will be taken considering PSTN line costs. In fact what VoIP does is the convert this:

Home Telephone1 --- (PSTN) --- Home Telephone2
PSTN great distance calling cost

into this:

      Home Telephone1 --- (PSTN) --- PC1   +
PC2 ---- (PSTN) --- Home Telephone2 =
2 PSTN short distance calling costs

To save money you need that:

2 PSTN short distance calling costs < PSTN great distance calling cost

Typically "short distance calling" refers to a "city cal" while "great distance calling" can be an "intercontinental call"!

Voip Setup

In this chapter we try to setup VoIP system, simple at first, then more and more complex.

Simple communication: IP to IP

       A (Sound card)   -  -  -    B (Sound card) - - - calls and viceversa.

A and B should have

  1. an application like Microsoft Netmeeting, Internet Switchboard, Openh323 (under Windows environment) or Ohphone, Gnomemeeting (under Linux), installed and properly configured.
  2. a network card or other kind of TCP/IP interface to talk each other.

In this kind of view A can make a H323 call to B (if B has server side application active) using B IP address. Then B can answer to it if it wants. After accepting call, VoIP data packets start to flow.

Using names

Under Microsoft Windows a NetBIOS name can be used instead of an IP address.

          A            -  -  -             B - - -

John - - - Alice

John calls Alice.

This is possible cause John call request to Alice is converted to IP calling by the NetBIOS protocol.

The above 2 examples are very easy to implement but aren't scalable.

In a more big view such as Internet it is impossible to use direct calling cause, usually, the callers don't know the destination IP address. Furthermore NetBIOS naming feature cannot work cause it uses broadcast messages, which typically don't pass ISP routers .

You can also use DNS to solve name in IP address: for example you can call ''box.domain.com''.

Internet calling using a WINS server

The NetBIOS name calling idea can be implemented also in a Internet environment, using a WINS server: NetBIOS clients can be configured to use a WINS server to resolve names.

PCs using the same WINS server will be able to make direct calling between them.

A (WINS Server is S) - - - - I  - - - -  B (WINS Server is S)
E - - - - - S (WINS Server)
C (WINS Server is S) - - - - R
E - - - - D (WINS Server is S)

Internet communication

A, B, C and D are in different subnets, but they can call each other in a NetBIOS name calling fashion. The needed is that all are using S as WINS Server.

Note: WINS server hasn't very high performance cause it use NetBIOS feature and should only be used for joining few subnets.

ILS server

ILS is a kind of server which allows you to solve your name during an H323 calling: when you start VoIP application you first register to ILS server using a name, then everyone will be able to see you using that name (if he uses same Server ILS!).

A big problem: the masquering.

A problem of few IPs is commonly solved using the so called masquering (also NAT, network address translation): there is only 1 IP public address (that Internet can directly "see"), the others machines are "masqueraded" using all this IP.

A - - -

B - - - Router with NAT - - - Internet

C - - -

This doesn't work

In the example A,B and C can navigate, pinging, using mail and news services with Internet people, but they CANNOT make a VoIP call. This because H323 protocol send IP address at application level, so the answer will never arrive to source (that is using a private IP address).


  • there is a Linux module that modifies H323 packets avoiding this problem. You can download the module here. To install it you have to copy it to source directory specified, modify Makefile and go compiling and installing module with "modprobe ip_masq_h323". Unfortunately this module cannot work with ohphone software at this moment (I don't know why).

A - - - Router with NAT

B - - - + - - - Internet

C - - - ip_masq_h323 module

This works

  • There is a application program that also solves this problem: for more see Par 5.7

A - - -

B - - - PhonePatch - - - Internet

C - - -

This works

Open Source applications

Ohphone Sintax

Sintax is:

"ohphone -l|--listen [options]"

"ohphone [options]... address"

  • "-l", listen to standard port (1720)
  • "address", mean that we don't wait for a call, but we connect to "address" host
  • "-n", "--no-gatekeeper", this is ok if we haven't a gatekeeper
  • "-q num", "--quicknet num", it uses Quicknet card, device /dev/phone(num)
  • "-s device", "--sound device", it uses /dev/device sound device.
  • "-j delay", "--jitter delay", it change delay buffer to "delay".

Also, when you start ohphone, you can give command to the interpreter directly (like decrease AEC, Automatic Echo Cancellation).


Gnomemeeting is an application using GUI interface to make call using VoIP. It is very simple to use and allows you to use ILS server, chat and other things.

Setting up a gatekeeper

You can also experiment gatekeeper feature


(Terminal H323) A - - -
(Terminal H323) B - - - D (Gatekeeper)
(Terminal H323) C - - -

Gatekeeper configuration

  1. Hosts A,B and C have gatekeeper setting to point to D.
  2. At start time each host tells D own address and own name (also with aliases) which could be used by a caller to reach it.
  3. When a terminal asks D for an host, D answers with right IP address, so communication can be established.

We have to notice that the Gatekeeper is able only to solve name in IP address, it couldn't join hosts that aren't reachable each other (at IP level), in other words it couldn't act as a NAT router.

You can find gatekeeper code here: openh323 library is also required.

Program has only to be launch with -d (as daemon) or -x (execute) parameter.

In addition you can use a config file (.ini) you find here.

Setting up a gateway

As we said, gateway is an entity that can join VoIP to PSTN lines allowing us to made call from Internet to a classic telephone. So, in addition, we need a card that could manage PSTN lines: Quicknet LineJack does it.

From OpenH323 web site we download:

  1. driver for Linejack
  2. PSTNGw application to create our gateway.

If executable doesn't work you need to download source code and openh323 library, then install all in a home user directory.

After that you only need to launch PSTNGw to start your H323 gateway.

Compatibility Matrix

First Matrix refers to:

  1. Software intercommunications (i.e. Netmeeting with SwitchBoard)
  2. Software/Driver/Hardware talking (i.e. Netmeeting can use a PhoneJACK card).

NetmeetingSwitchBoardSimph323OhPhoneLinPhoneSpeak-FreelyHW PhoneJACKHW LineJACK

Second Matrix refers to Gateway softwares that manage LineJACK card.

| |HW LineJACK GW| SwitchBoard | PSTNGW |
|HW LineJACK GW| _ | V | V |
| SwitchBoard | V | _ | _ |
| PSTNGW | V | _ | _ |


  • V : Works
  • X : Doesn't Work
  • -- : Doesn't care

Voip Card Setup

Here we see how to configure special hardware card in Linux and Windows environment.

Quicknet PhoneJack

As we saw, Quicknet Phonejack is a sound card with VoIP accelerating capability. It supports:

  • G.711 normal and mu/A-law, G.728-9, G.723.1 (TrueSpeech) and LPC10.
  • Phone connector (to allow calling directly from your phone) or
  • Mic & speaker jacks.

Quicknet PhoneJack is a ISA (or PCI) card to install into your Pc box. It can work without an IRQ.

Software installation

Under Windows you have to install:

  1. Card driver
  2. Internet Switchboard application (working only with Internet, using newer Quicknet cards)

all downloadable from Quicknet web site

After Switchboard has been installed, you need to register to Quicknet to obtain full capability of your card.

When you pick up the phone Internet Switchboard wakes up and waits for your calling number (directly entered from your phone), you can:

  1. enter an asterisk, then type an IP number (with asterisks in place of dot) with a # in the end
  2. type directly a PSTN phone number (with international prefix) to call a classic phone user. In this case you need a registration to a gateway manager to which pay for time.
  3. enter directly a quick dial number (up to 2 digits) you have previously stored which make a call (IP or PSTN).

Internet Swichboard is h323 compatible, so if you can use, for example, Microsoft Netmeeting at the other end to talk.

Warning!! Internet Switchboard NEED to be connected to Internet when used with newer Quicknet cards

In place of Internet Switchboard you can use openh323 application openphone (using GUI) or ohphone (command line).

Under Linux you have to install:

  1. Card driver, from Quicknet web site. After downloaded you have to compile it (you must have a /usr/src/linux soft or hard link to your Linux source directory): type make for instructions.
  2. Application openphone or ohphone.
  3. If you are a developer you can use SDK to create your own application (also for Windows).


With Internet Switchboard (and with other application) you can:

  1. Change compression algorithm preferred
  2. Tune jitter delay
  3. Adjust volume
  4. Adjust echo cancellation level.

Quicknet LineJack

This card is very similar to the previous, it supports also gateway feature.

We only notice that we have to download PSTNGx application (for Linux and Windows) or we use Internet Switchboard to gateway feature.

VoiceTronix products

  1. First download software here
  2. Untar it
  3. Modify 'src/vpbreglinux.cpp' according to file README
  4. type 'make'
  5. type 'make install'
  6. cd to src
  7. type 'insmod vpb.o'
  8. retrieve (from console of from 'dmesg' output command) major number, say MAJOR
  9. type 'mknod /dev/vpb0 c MAJOR 0' where MAJOR is the above number
  10. cd to unittest and type './echo'

Follow README file for more help.

I personally haven't tested VoiceTronix products so please contact VoiceTronix web site for support.


Hardware requirement

To create a little VoIP system you need the following hardware:

  1. PC 386 or more
  2. Sound card, full duplex capable
  3. a network card or connection to internet or other kind of interface to allow communication between 2 PCs

All that has to be present twice to simulate a standard communication. The tool above are the minimal requirement for a VoIP connection: next we'll see that we should (and in Internet we must) use more hardware to do the same in a real situation. Sound card has be full duplex unless we couldn't hear anything while speaking! As additional you can use hardware cards (see next) able to manage data stream in a compressed format (see Par 4.3).

Hardware accelerating cards

We can use special cards with hardware accelerating capability. Two of them (and also the only ones directly managed by the Linux kernel at this moment) are the

  1. Quicknet PhoneJack
  2. Quicknet LineJack
  3. VoiceTronix V4PCI
  4. VoiceTronix VPB4
  5. VoiceTronix VPB8L

Quicknet PhoneJack is a sound card that can use standard algorithms to compress audio stream like G723.1 (section 4.3) down to 4.1 Kbps rate.

It can be connected directly to a phone (POTS port) or a couple mic-speaker.

It has a ISA or PCI connector bus.

Quicknet LineJack works like PhoneJack with some addition features (see next).

VoiceTronix V4PCI is a PCI card pretty like Quicknet LineJack but with 4 phone ports

VoiceTronix VPB4 is a ISA card equivalent to V4PCI.

VoiceTronix VPB8L is a logging card with 8 ports.

For more info see Quicknet web site and VoiceTronix web site

Hardware gateway cards

Quicknet LineJack and VoiceTronix cards can be connected to a PSTN line allowing VoIP gateway feature.

Then you'll need a software to manage it (see after).

Software requirement

We can choose what O.S. to use:

  1. Win9x
  2. Linux

Under Win9x we have Microsoft Netmeeting, Internet Phone, DialPad or others or Internet Switchboard (from Quicknet web site) for Quicknet cards.

Warning!!: Latest Quicknet cards using Swithboard (older version too) NEED to be connected to Internet to get working for managing Microtelco account (not free of charge), so if you plan to remain isolated from Internet you need to install OpenH323 software.

For VoiceTronix cards you can find software at VoiceTronix web site

Under Linux we have free software GnomeMeeting, a clone of Microsoft Netmeeting, while in console mode we use (also free software) applications from OpenH323 web site: simph323 or ohphone that can also work with Quicknet accelerating hardware.

Attention: all Openh323 source code has to be compiled in a user directory (if not it is necessary to change some environment variable). You are warned that compiling time could be very high and you could need a lot of RAM to make it in a decent time.

Gateway software

To manage gateway feature (join TCP/IP VoIP to PSTN lines) you need some kind of software like this:

  • Internet SwitchBoard (only when connected to Internet) for Windows systems also acting as a h323 terminal;
  • PSTNGw for Linux and Windows systems you download from OpenH323.

Gatekeeper software

You can download as gatekeeper the Free product Openh323 Gatekeeper (GK) from here.

Version 2.0 of it supports "proxy function" to enabe talking from/to a private network.

Other software

In addition I report some useful software h323 compliant:

  • Phonepatch, able to solve problems behind a NAT firewall. It simply allows users (external or internal) calling from a web page (which is reachable from even external and internal users): when web application understands the remote host is ready, it calls (h323) the source telling it all is ok and communication can be established. Phonepatch is a proprietary software (with also a demo version for no more than 3 minutes long conversations) you download from here.

Same function can be obtained using "Proxy" function of Gatekeeper Gnugk (see before).

Voip Overview

What is VoIP?

VoIP stands for Voice over Internet Protocol. As the term says VoIP tries to let go voice (mainly human) through IP packets and, in definitive through Internet. VoIP can use accelerating hardware to achieve this purpose and can also be used in a PC environment.

How does it work?

Many years ago we discovered that sending a signal to a remote destination could have be done also in a digital fashion: before sending it we have to digitalize it with an ADC (analog to digital converter), transmit it, and at the end transform it again in analog format with DAC (digital to analog converter) to use it.

VoIP works like that, digitalizing voice in data packets, sending them and reconverting them in voice at destination.

Digital format can be better controlled: we can compress it, route it, convert it to a new better format, and so on; also we saw that digital signal is more noise tolerant than the analog one (see GSM vs TACS).

TCP/IP networks are made of IP packets containing a header (to control communication) and a payload to transport data: VoIP use it to go across the network and come to destination.

Voice (source) - - ADC - - - Internet - - - DAC - - Voice (dest)

What is the advantages using VoIP rather PSTN?

When you are using PSTN line, you typically pay for time used to a PSTN line manager company: more time you stay at phone and more you'll pay. In addition you couldn't talk with other that one person at a time.

In opposite with VoIP mechanism you can talk all the time with every person you want (the needed is that other person is also connected to Internet at the same time), as far as you want (money independent) and, in addition, you can talk with many people at the same time.

If you're still not persuaded you can consider that, at the same time, you can exchange data with people are you talking with, sending images, graphs and videos.

Then, why everybody doesn't use it yet?

Unfortunately we have to report some problem with the integration between VoIP architecture and Internet. As you can easy imagine, voice data communication must be a real time stream (you couldn't speak, wait for many seconds, then hear other side answering): this is in contrast with the Internet heterogeneous architecture that can be made of many routers (machines that route packets), about 20-30 or more and can have a very high round trip time (RTT), so we need to modify something to get it properly working.

In next sections we'll try to understand how to solve this great problem. In general we know that is very difficult to guarantee a bandwidth in Internet for VoIP application.


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