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  • About VoIP
    • What is VoIP and what it can do for you
    • Introduction to VoIP (video)
    • Why should you switch to VoIP services?
    • Analog Telephony
    • Digital Telephony
    • What is SIP?
    • How to start with VoIP telephony
    • Web based VoIP
    • How to choose a right VoIP provider?
    • Wi-Fi network and VoIP
    • VoIP Codecs
    • Free sip account
    • Confidential calls
    • VPN: UDP or TCP?
    • Mobile VoIP
    • VoIP on your mobile
  • Asterisk IP-PBX
    • All about Asterisk
    • About Mark Spencer
    • Asterisk SIP Media NAT
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    • Cisco ATA186 + Asterisk Config
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Cisco IP phone

Plain explain: IP-phone

By Nik On February 27, 2013 · In Cisco, SIP protocol, SIP termination, VoIP calls quality

We all strive to get a quality telephony, but when we do some analyze of the voice quality issues and most common reasons that may lead to the problems with the VoIP service quality we should take into account the VoIP equipment, including IP-phones.

Looking closer: what is an IP-phone

VoIP phone is a [...]

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VoIP SIP termination providers

Plain explain: What is SIP termination and what is SIP origination?

By Nik On February 26, 2013 · In SIP termination, VoIP service

On the telecom market many (if not all) TELCO companies (including SIP providers of course) offer services that is based on  VoIP. Almost all of them may provide a SIP termination and SIP origination. In this short article I will try to define these terms (SIP termination and SIP origination) and briefly explain what is [...]

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us-market_voip-sip

US fixed VoIP market to see steady growth

By Nik On January 11, 2013 · In VoIP industry news

Those who have been paying attention to voice communications trends over the past couple of years should not be surprised by the numbers from the recent TechNavio report regarding fixed VoIP services in the United States. The company said that between 2012 and 2016, the market will grow at a rate of about 10.15 %per year.

“One of [...]

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economy_voip-sip

Save money with VoIP and unified communications

By Nik On January 11, 2013 · In VoIP industry news

While travel and procurement costs have been rising for many aspects of business, such as travel and procurement of certain types of technology, Mini Swamy, a TMCnet contributor, wrote that VoIP and unified communications can help companies save some money. She looked at a report by Digitalolympus.com, which said the right implementation of these programs can help companies of [...]

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small-business_voip-sip

IP telephony can help level the playing field for small businesses

By Nik On January 11, 2013 · In IT news, VoIP industry news

For years, small companies have been tied down by carrier-based phone systems that are expensive and cumbersome. The Guardian said there are ways to help small businesses compete with big businesses and one way to do that is by adopting an IP telephony system in lieu of phone lines.

“While a business may be based in, say, [...]

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no-phone_voip-sip

The loss of a landline means big change in communications

By Nik On January 11, 2013 · In VoIP industry news

The loss of a landline is something that every business will have to deal with over the next few years. A recent report by inetwork, a division of ATLANTIC-ACM, shows that 74 percent of those polled believe that the death of the office landline and deskphone will be one of the most significant forces in [...]

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ubuntu-linux-phone

Ubuntu (Linux) on your phone? Yes! And now officially!

By Nik On January 4, 2013 · In Open source, Ubuntu

News from Canonical: Ubuntu is coming to the phone

When we began developing Unity a few years ago, the aim was to create a single family of interfaces that work the same way on different devices. This means that unlike most of our rivals, we are able to use a single underlying OS across all [...]

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facetime_150x150

QoS For FaceTime, bandwidth requirements and Firewall config

By Nik On January 3, 2013 · In Apple, FaceTime, Prioritization and traffic shaping, QoS - Quality of Service, VoIP calls quality

Facetime – Calls Quality, QoS and firewall ports in use

Video calling will become an increasingly widespread form of communication in coming years, moving from
broad adoption in the consumer market to selective endorsement in the corporate world. The same executives
who video-call their families when on the road will also use FaceTime [...]

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Merry Christmas & Happy New Year 2013

Merry Christmas and Happy New Year!

By Nik On December 25, 2012 · In Non VoIP news

Dear Friends, Readers, Writers and all colleagues of VoIP-SIP.ORG,

We would like to thank all those with whom we have done some work on Open Source projects. A lot of stuff was done at this year and it was great to work together with you guys!

Merry Christmas and Happy New Year! Stay crazy like [...]

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How to config axvoice

How to Configure Axvoice Equipment?

By Nik On December 11, 2012 · In VoIP service

INTRODUCTION TO AXVOICE:

Axvoice is a well known low cost VoIP phone service provider that provides its services in USA and Canada. Axvoice offers a huge variety of residential and business plans. Pay as you go is their cheapest plan ($4.99/month). This plan allows you to pay only when you need to call. $8.25/month is [...]

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what is voip termination - how these calls works?

Cloud VoIP vs. on-premise VoIP: Choosing the right one for your business

By Nik On December 6, 2012 · In Cloud VoIP

As VoIP continues its growth in the business market, more and more companies are finding out that selecting the right type of phone solution sometimes can be a bit confusing.  This is because there are multiple options within the VoIP phone market.  One of the most common areas of misunderstanding is the distinction between on-premise [...]

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professor jrgleuthold

26 terabits per second data transmission achieved

By Nik On December 4, 2012 · In IT news

With video content consuming ever more bandwidth, the need for faster data transmission rates has never been greater. Now a team of scientists at Germany’s Karlsruhe Institute of Technology (KIT) are claiming a world record in data transmission with the successful encoding of data at a rate of 26 terabits per second on a [...]

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Ekiga_in_a_Call3

A fully functional VoIP Client (SIP) finally released (free download)

By Nik On November 29, 2012 · In Open source, Softphone VoIP

Great news for today, guys! The legendary Linux softphone is back for more (available for Windows also)!

Three years after the 3.2 release, Ekiga 4.0 aka “The Victory Release” is finally available.

This is a major release with many major improvements.

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cisco-voip2

Fundamentals of SIP from Cisco :-)

By Nik On August 23, 2012 · In Cisco, SIP protocol

Session Initiation Protocol, SIP, is poised to continue its reshaping of your collaboration and communication network. Hope you will enjoy watching this short, but very good video below!

This Fundamentals animation was produced as part of a TechWiseTV episode 66 ‘SIP, Session Management and Beyond. Just for the reference.

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webrtc_2

WebRTC from Google: making real-time communication free to implement

By Nik On July 12, 2012 · In Android, Google Voice, Open source

After Microsoft bought Skype for US$8.5 billion, Google has released a developer preview of WebRTC – an open framework enabling implementation of voice and video Real Time Communications in the browser with the use of HTML 5 and JavaScript APIs.

It’s possible that the Skype acquisition could mean restricting the client from technologies and devices [...]

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questions

What is VoIP termination

By Nik On July 11, 2012 · In Asterisk, SIP protocol

Many people keep asking me – what is VoIP Termination?

There is an “official” answer: VoIP Termination is the process of routing a VoIP telephone call (or calls flow) from one telephone service provider to another. What is that meant? Ok, lets imagine you have some phone lines (IP-phones or VoIP gateways with regular phones [...]

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compass_25547_voip_course

Introduction to Voice over IP (VoIP)

By Nik On July 11, 2012 · In Asterisk, Mobile VoIP, Open source, SIP protocol

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 [...]

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linux_version_check

Linux: how to check OS version installed

By Nik On June 18, 2012 · In VoIP calls quality

Sometimes it’s a pain in the ass to install software on unix based systems without having prior knowledge to the OS/kernel versions. You’d do something on debian that doesn’t work on CentOS, Fedora has yum pre-installed where as RHEL4 comes with up2date but you have to have a key, then there’s always RPM’s but who [...]

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Hey!

By Nik On June 13, 2012 · In Asterisk

To all people who has an experience with Asterisk and Linux – quote of the day: “I have not failed. I’ve just found 10,000 ways that won’t work!” by Thomas Edison. Keep trying guys and you’ll find how to do it right!

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free phone calls

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

By Nik On March 28, 2012 · In Google Voice, Mobile VoIP, Open source, SIP protocol, Symbian, VoIP calls quality

In this article we will talk about Nokia Internet telephony and SIP settings since many people asking me about that. I have check all configuration myself on my old Nokia E52 as well as E71 buddy and everything works fine. Therefore you can use the same setup procedure on any E-series with Symbian.

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TCP vs UDP – you must know this

By Nik On December 12, 2011 · In Prioritization and traffic shaping, SIP protocol, VoIP calls quality

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 [...]

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Easy setup how to Asterisk with Google talk and Google voice

Google voice / Google talk and Asterisk configuration

By Nik On December 8, 2011 · In Asterisk, Google Voice, Open source

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 [...]

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VoIP or IP Telephony?

By Nik On November 15, 2011 · In SIP protocol

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?

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Asterisk 10.0.0-rc1 Now Available!

By Nik On November 14, 2011 · In Asterisk, Open source, SIP protocol

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 [...]

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SIP VoIP calls NAT

SIP based VoIP behind NAT

By Nik On November 10, 2011 · In Asterisk, Google Voice, History, Open source, SIP protocol

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 [...]

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← Previous Entries
  • Recent articles

    • Plain explain: IP-phone
    • Plain explain: What is SIP termination and what is SIP origination?
    • US fixed VoIP market to see steady growth
    • Save money with VoIP and unified communications
    • IP telephony can help level the playing field for small businesses
    • The loss of a landline means big change in communications
    • Ubuntu (Linux) on your phone? Yes! And now officially!
    • QoS For FaceTime, bandwidth requirements and Firewall config
    • Merry Christmas and Happy New Year!
    • How to Configure Axvoice Equipment?
    • Cloud VoIP vs. on-premise VoIP: Choosing the right one for your business
    • 26 terabits per second data transmission achieved
    • A fully functional VoIP Client (SIP) finally released (free download)
    • Fundamentals of SIP from Cisco :-)
    • WebRTC from Google: making real-time communication free to implement
    • What is VoIP termination
    • Introduction to Voice over IP (VoIP)
    • Linux: how to check OS version installed
    • Hey!
    • How to configure Internet tel. and SIP settings on Nokia phone (E52)
    • TCP vs UDP – you must know this
    • Google voice / Google talk and Asterisk configuration
    • VoIP or IP Telephony?
    • Asterisk 10.0.0-rc1 Now Available!
    • SIP based VoIP behind NAT
    • VoIP Quick start guide
    • History: Kellogg Field Phone (World War I)
    • Running VoIP via VPN (SSL) – voice quality
    • Skype App directory
    • G.711: u-law or a-law?
    • Wi-Fi access point/router optimization for VoIP and other real time apps
    • Agri-Cube grows mass quantities of vegetables in a one-car parking spot
    • Quick Comparison of freeware IP-PBX platforms: Asterisk vs Open SER
    • Microsoft enabling the ability to eavesdrop on VoIP conversations
    • Nimbuzz growing even without Skype
    • Skype protocol hack
    • Call DSN number
    • How to make your VoIP calls private and confidential
    • Think on solutions: VoIP phone system
    • Asterisk and Google Voice
    • VoIP Codec: Payload size
    • Nokia SIP settings
    • The QoS Dilemma
    • VoIP calls over satellite links
    • VoIP for Facebook!
    • VoIP client behind a VPN with DD-WRT
    • Digital Telephony
    • Analog Telephony
    • Mobile VoIP – the future of mobile communications
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    • Multiplexing RTP Data and Control Packets on a Single Port
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  • Plain explain: IP-phone
    We all strive to get a quality telephony, but when we […]

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