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Introduction to Voice over IP (VoIP)

March 29, 2012 by nicom | 0 comments

One important step into adopting VoIP is to choose a VoIP service, which will allow you to make and receive cheap or free local and international phone calls. It is important to choose the right type of VoIP service. Your needs and the way you will communicate should help you decide which type of VoIP service to choose. The list below shows the existing types of VoIP service, and helps you decide which type suits you best. Continue Reading →

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How to configure Internet tel. and SIP settings on Nokia phone (E52)

March 28, 2012 by nicom | 2 Comments

This article about Nokia Internet tel. and SIP settings. I have check it on my Nokia E52 and everything works fine. Therefore you can use the same setup procedure on any E-series with Symbian. Here we go:

 

1. Download and install SIP Voip Settings Tool for Nokia

This application adds missing VoIP functionality to your E52, you can download it from Nokia site: select “SIP VoIP 3.x Settings (164 kB)” option and press “Download” button. This will download SIP_VoIP_3_x_Settings_v2_0_en.sis file.

Note: Nokia site may require you logged in before allowing the download. If this happens, you can register as a developer using “Join” link (it is free) or you may send me a message if you will have any questions on this.

After you receive .sis file, transfer it to your mobile with e.g. USB cable and run it from File Manager. This will install the application. After the program is installed, reboot (power cycle) your phone, go to Menu > Ctrl.panel and check if “Net Settings” application has been added to the list.

2-A. Create a SIP profile in your Nokia phone (simple way)

Go to Menu > Ctrl.panel, run Net settings application, go to Advanced VoIP settings > Create new service then select Create new SIP profile option. Continue Reading →

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TCP vs UDP – you must know this

December 12, 2011 by nicom | 2 Comments

I think TCP is an overused protocol and  I think that UDP is an underused protocol.

This is an argument I’ve been having quite a bit with people lately, so I’ve decided i’ll lay out my reasoning here so I don’t have to type or recite it at people over and over. Understanding how TCP works and how UDP works and know the difference is very important for VoIP engineer or any other person who believe that VoIP is a future of all International, domestic and local telephony systems. It must be quality voice and you must know how it works and how to make it’s quality better or at least how this task can be done…

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Easy setup how to Asterisk with Google talk and Google voice

December 8, 2011
by nicom
0 comments

Google voice / Google talk and Asterisk configuration

This article describes the configuration process of Asterisk with Google voice.

Asterisk communicates with Google Voice and Google Talk using the chan_gtalk Channel Driver and the res_jabber Resource module. Before proceeding, please ensure that both are compiled and part of your installation. Compilation of res_jabber and chan_gtalk for use with Google Talk / Voice are dependant on the iksemel library files as well as the OpenSSL development libraries presence on your system. Continue Reading →

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VoIP or IP Telephony?

November 15, 2011 by nicom | 0 comments

What is The Difference Between VoIP and IP Telephony?

Most people, including consumers, use the terms VoIP (Voice over IP) and IP Telephony interchangeably, equating one to the other. But what’s the difference between the two? Continue Reading →

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Asterisk 10.0.0-rc1 Now Available!

November 14, 2011 by nicom | 0 comments

Digium releases Asterisk 10.

As you may already know Asterisk is a communications platform that allows developers to create powerful business phone systems and unified communications solutions. Since its introduction 12 years ago, Asterisk has been used, free of charge, in nearly every country of the world to power telephone and other communications systems. It has been downloaded millions of times, including two million last year alone, establishing Asterisk as the most popular open source telephony engine.

The most important new feature in Asterisk 10 is its wide-band media engine. Digium has replaced Asterisk’s telephony-grade media engine with a more advanced one, providing support for studio-quality audio and a nearly unlimited number of codecs. By supporting high and ultra high-definition voice, Asterisk can now be used to power communications applications that would have otherwise required specialized or expensive equipment and service in order to convey nuances in speech or emotion. Digium has also updated Asterisk’s media support for Asterisk 10, with several new codecs, including Skype’s SILK codec, 32kHz Speex support and pass-through support for CELT.

Built with open source community support
Continue Reading →

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SIP based VoIP behind NAT

November 10, 2011 by nicom | 0 comments

SIP VoIP calls NAT

For all the technology behind Voice over IP (VoIP), you’d expect that it would work on every network, but this unfortunately isn’t the case. Network Address Translation (NAT) is a common practice used in networks, and it doesn’t play well with VoIP. Solving this problem requires an understanding of NAT, VoIP and your VoIP setup. This article focuses on the SIP protocol for VoIP and the Asterisk VoIP software, but the problems and solutions are applicable to most other situations. Continue Reading →

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VoIP Quick start guide

November 9, 2011 by nicom | 0 comments

Hi there. Today I found a quick start guide on VoIP written for Cisco 3600 series modular routers- click here to download PDF. I think it could be useful even if you just a newbie and what to learn how VoIP works, so I decided to share it. Please let me know if you think it’s useful and please don’t forget to support us and click on advertising (right column). Continue Reading →

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History: Kellogg Field Phone (World War I)

October 19, 2011 by nicom | 0 comments

Kellogg WWI Field Phone

Ask people familiar with telecommunications history and they’ll tell you that the first text message sent from a mobile phone was sent in 1993. But there was  a phone that sent text messages more than 75 years before that! Amazing, bu the Kellogg Switchboard Supply Company field phone, model 1917 (and we’ve also seen model EE 3) provided American soldiers in WWI a portable telephone and telegraph communications in a single box.

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Running VoIP via VPN (SSL) – voice quality

October 18, 2011 by nicom | 1 Comment

The protocol overhead caused by the encapsulation of VoIP protocol within VPN dramatically increases the bandwidth requirements for VoIP calls, thus making the VoIP over VPN protocols too “fat” to be used over a mobile data connections like GPRS, EDGE or UMTS. Although VoIP over VPN is not as usable in mobile environments, it is sometimes used to create “encrypted VoIP trunk” between different sites of a corporations, running VoIP PBX interconnections over a VPN connection.

VoIP is often written off as an application that will not work well over an SSL VPN link. To test that argument, we have discovered ten VPN products (SSL) in four network scenarios to see how well VoIP calls were handled in this conditions.

The news is generally good. In high-bandwidth, low-latency environments, there is virtually no difference in quality between an unencrypted VoIP call and the same call made over an SSL VPN.

Even better news is our discovery that a VoIP call made over SSL VPN on a typical broadband Internet connection is of higher quality than an unencrypted call. The only bad news comes with truly awful network connections: ones with high loss and limited bandwidth. In this environment, neither unencrypted VoIP calls nor SSL VPN-protected calls will be considered acceptable, below MOS of 3 (Mean Opinion Score).

While our results do show some differences between products, small variations in the MOS should not be considered significant. What is more important, our testing demonstrates that SSL VPN and VoIP work together well over broadband networks, even in the face of some network loss and congestion. Continue Reading →

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Skype App directory

August 31, 2011 by nicom | 0 comments

Many people using Facebook, Apple and Android applications. Skype has now introduced their applications from third party vendors that has to  enhance usability.

Looks like after Microsoft purchased  Skype, it’s on a roll. They bought several related tech companies in the last few months and they are ready to enhance their VoIP offering by allowing third party companies to build applications on top of Skype. Although, the add-ons were available in the market for a long time, there wasn’t a official App directory untill now.

Skype Applications directory works almost the same  like Apple App Store. Continue Reading →

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G.711: u-law or a-law?

August 23, 2011 by nicom | 1 Comment

As you already know G.711 is a high quality voice codec that we have support for in Asterisk as well as in  many other open source and commercial VoIP platforms. G.711 uses logarithmic PCM (pulse code modulation), a standard as old as from 1972. G.711 is pretty much the norm for IP-telephony where there is enough bandwidth, hence most IP-phones will come preset to this codec (either u-law or a-law, see below). Whether you can use it or not, depends on your available bandwidth (with IP headers it’s bandwidth requirements close to 80 Kbit/s per each active line). If you have a normal broadband connection of cheapest type, you can still get at least two phone calls going with this codec (or one call and one on hold with music).

Watch the upstream speed on ADSL (and satellite links as well), this is the limiting factor when you decide what codec to use.

Now we will try to understand what is the difference between µ-law and a-law versions of G.711 voice codec and which one is better for your particular setup. Continue Reading →

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Wi-Fi access point/router optimization for VoIP and other real time apps

August 23, 2011 by nicom | 1 Comment

In order to improve the bandwidth allocation for certain apps in your wireless network you need to enable QoS (quality of service) feature on your router/access point. Please make sure that you have enough bandwidth available (down and up) on your Internet link.

If the quality of your VoIP calls or Skype calls/video start breaking up, or your favorite on-line radio stream gets clogged whenever your wife/child  starts download new movie using BitTorrent ( P2P) or playing World of Warcraft, you may improve the performance without adding more bandwidth (higher service package). Most LAN and Wi-Fi routers sold in the past few years have a quality-of-service (QoS) feature.

As an example I will use Linksys router as I have WRT54GL at home, but other models or vendors will have pretty much the same fields that we will need to setup. Continue Reading →

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Quick Comparison of freeware IP-PBX platforms: Asterisk vs Open SER

August 10, 2011 by nicom | 0 comments

Ok, most of us  know that Voice Internet Protocol (VoIP) Telephony refers to the technology used for making telephone calls over the Internet. We trying to make this technology as easy as possible for engineers/geeks who trying to deploy it all around and also for users who just want to pick up the phone and make a call.


TollFreeForwarding.com

I think it’s very important to understand the complete call flow in order to be ready to solve some issues that could be here and there with VoIP. Nowadays there are  two major platforms  used to implement VoIP telephony: one is a great Asterisk (thanks to Mark Spencer, cheers!) and OpenSER. In this post we will try to compare Asterisk and OpenSER side by site and understand what is the main differences between these two.

Prelude

If you work with IP telephony, it’s quite possible that you have not heard about OpenSER, but certainly you must have heard about Asterisk. Continue Reading →

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July 13, 2011
by nicom
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Microsoft enabling the ability to eavesdrop on VoIP conversations

I’ve reading an interesting article on eweek.com that describes the problems that large companies face as they try to diversify – specifically the move of Microsoft from software towards IP telephony.

News Analysis: Now that Microsoft is on its way to becoming a telephone company, it is finding itself subject to a lot of things that a mere software company never had to deal with.

“As eWEEK’s Fahmida Rashid explains, Microsoft filed a patent in 2009 for technology that would greatly simplify the process of monitoring a VOIP conversation. At the time it was filed, this patent got little attention. After all, while Microsoft had telephony products at the time, it wasn’t a carrier. So if Microsoft had a need (or a warrant) that required listening in to a conversation over VOIP on its own phone system, it wouldn’t have been that hard to arrange. Continue Reading →

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June 24, 2011
by nicom
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Nimbuzz growing even without Skype

According to Nimbuzz, since July of 2010, they have more than 28 million new users joined their network. That amounts equals to 100,000 new peoples for every 24 hours and they’ve doubled in size in year to 50 million users.

Even with Skype integration removed, Nimbuzz still supports Facebook, Yahoo, Windows Live, Google Talk, AIM, and others. So it’s still a popular mobile VoIP and instant messaging app that I am using myself.

It does a good job of consolidating all your various IM / VoIP networks into a single mobile application. I do hope that Microsoft, the new owners of  Skype, will change their minds and allow apps like Nimbuzz or fring (who also pushed by Skype to overboad) to connect to the Skype network.

Source: Nimbuzz Blog

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June 24, 2011
by nicom
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Skype protocol hack

Skype protocol hack could have been prevented claims StarForce

StarForce’s comments come in the wake of blog postings by security researcher Efim Buchmanov who, earlier this month, claimed to have reverse engineered the Skype protocol.

“My aim is to make skype open source. And find friends who can spend many hours for completely reverse it.” he posted on his blog recently adding links to download executable files compatible with Skype versions 1.4, 3.8, and 4.1: Continue Reading →

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June 23, 2011
by nicom
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Call DSN number

What Organizations Use the DSN Network?

The majority of Department of Defense (DoD) commands have one or more phone lines in the DSN network. Just because a command has a DSN phone number, however, does not mean that the number(s) will automatically be listed in the global DSN directory. The command has to formally request via the Defense Information Systems Agency (DISA) to be listed in the directory.

How to dial a DSN Number

Although the DSN phone network can be accessed by a commercial phone at a command or agency that is an authorized network user, they can not be called from a personal or home phone.

First of all you need to know the access numbers to dial on your phone to access the DSN network. Typically this will be a one or two digit combination such as “88,” “89”, or “99.” Then dial the DSN area code for the geographic region that you are calling. If you are dialing a DSN phone number for the same region that you are located, this step can be omitted.

The DSN area codes are:

Continental US/Puerto Rico – 312, 502 for video calls
Europe – 314, 504 for video calls
Pacific – 315, 505 for video calls
Alaska – 317, 507 for video calls
Central Command (Southwest Asia)- 318, 504 for video calls
Canada – 319, 509 for video calls

Dial the seven digit DSN phone number and complete the phone call as you would with a commercial phone line.

If you need to make a secure phone call, you will need to follow the procedures at your command or organization to initiate a secure call after the person you are calling answers the phone.

Please note: you can’t direct dial a DSN number from a regular phone (PSTN) unless they have a regular phone number as well.

For example, many DSN numbers like 312-381-xxxx are 703-882-xxxx on a regular phone so if you needed to call 312-381-1234 on the DSN, you’d call 703-882-1234 on a regular phone. Definitely not all DSN numbers work that way but some do. Other than that, you’d have to get someone to transfer your call for you.

You can try this number 719-567-1110 (to access DSN switchboard in Schriever AFB  Colorado Springs center) but I don’t know if they’ll let a civilian on the DSN transferred or not.

Also you may try convert a DSN number to its commercial equivalent and vice versa using these tables. Not all DSN numbers has an equivalent of public line.

Complete DSN directory is here. Hope it helps to get in touch with the person you want to call.

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June 21, 2011
by nicom
1 Comment

How to make your VoIP calls private and confidential

There is no doubt a chance exists that your IP phone calls can be tapped and listened to. We will discuss here how to make sure your calls stay private and secure.

The possibility your VoIP calls being monitored is much higher then of conventional phone calls. The reason for this is that IP telephony is not a part of the public phone network, where tapping into the communication line involves a specific, forcible wired link. Calls carried via the Internet, LAN, or a WAN transport can be intercepted merely by obtaining and examining the data packets by anybody with a communications protocol analyzer. Subsequently, virtually anyone can spy on your private as well as business VoIP activities – your rivals, business partners, employers and employees, state officials and law-enforcement agencies.

The best way to achieve reasonable security of your VoIP communications is to employ data encryption. Continue Reading →

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June 21, 2011
by nicom
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Think on solutions: VoIP phone system

It is often taken for granted that installing VoIP solutions will save customers time and money, but in actual fact getting Return on Investment depends on a number of factors.

Firstly if you are planning to use VoIP phone system for business purposes, think about how it will integrate into your existing Public Switched Telephone Network (PSTN) and/or GSM mobile network.

Using VoIP to VoIP calling is a low cost option but in cases where you may have to call a regular landline it might not work out cheaper. Therefore you should work out what you want your IP-PBXto do before ripping out all the existing cabling.

However daunting the installation and management of your calling system can seem it is actually possible to use bundled VoIP to customise your experience. Many VoIP companies offer 24 hour support and there is no need to waste hours on training for what is an easy to use interface.

The real key with VoIP is having a fast and reliable internet connection. With really fast broadband VoIP can actually deliver BETTER call quality than traditional analogue phones, but conversely a poor connection can create all sorts of problems.

Importantly whether you actually save money on VoIP depends on how expensive your broadband is. Normally it should save you money because you are getting two services (calling and surfing) for the price of one but it all depends on the package so shop around.

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