questions

What is VoIP termination

By 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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Introduction to Voice over IP (VoIP)

By 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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Hey!

By 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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Easy setup how to Asterisk with Google talk and Google voice

Google voice / Google talk and Asterisk configuration

By 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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Asterisk 10.0.0-rc1 Now Available!

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

By On August 23, 2011 · In Asterisk, Google Voice, Mobile VoIP, QoS - Quality of Service, VoIP calls quality

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 [...]

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Microsoft enabling the ability to eavesdrop on VoIP conversations

By On July 13, 2011 · In Asterisk, Mobile VoIP, SIP protocol

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 [...]

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Nimbuzz growing even without Skype

By On June 24, 2011 · In Android, Asterisk, Mobile VoIP, SIP protocol, VoIP calls quality, VoIP over VPN

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 [...]

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How to make your VoIP calls private and confidential

By On June 21, 2011 · In Asterisk, Mobile VoIP, Prioritization and traffic shaping, SIP protocol, VoIP calls quality, VoIP over VPN

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 [...]

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Think on solutions: VoIP phone system

By On June 21, 2011 · In Android, Asterisk, Mobile VoIP, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over VPN

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 [...]

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Asterisk and Google Voice

By On June 21, 2011 · In Asterisk, Google Voice, Mobile VoIP, Open source, SIP protocol

Asterisk: How to make calls using Google Prerequisites

Asterisk can communicate with Google Voice and Google Talk  using the chan_gtalk Channel Driver and the res_jabber  Resource module. Please ensure that both are compiled and part  of your installation prior following the configuration guide  below. Compilation of res_jabber and chan_gtalk for use with  Google [...]

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VoIP Codec: Payload size

By On June 16, 2011 · In Asterisk, Mobile VoIP, SIP protocol, VoIP calls quality

The size of the payload of each encoded voice packet influences two things: lag and bandwidth.

Every encoded packet that is sent incurs fixed bandwidth overheads (due to IP and other headers added to the data in the network). Thus, larger payloads incur a proportionately smaller overhead, thus reducing the nominal bandwidth utilisation. However, by [...]

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Nokia SIP settings

By On June 16, 2011 · In Android, Asterisk, Mobile VoIP, SIP protocol, VoIP calls quality, VoIP over VPN

Nokia setup for SIP-based VoIP service

UPDATE: New article about SIP settings (Internet tel.) for Nokia E52 is here

Now you can enjoy crystal-clear phone calls over the Internet using the new Nokia S60 phones. If you have access to a Wi-Fi or 3G connection, you can [...]

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VoIP calls over satellite links

By On June 14, 2011 · In Asterisk, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality

VoIP services via high latency satellite links: overview

As we all know the VoIP market expanding rapidly over the past few years and SIP based services allows the voice communication over global Internet network, giving the customers great savings and more convenient service.  In the past with cooper wires you can do nothing about bad [...]

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VoIP client behind a VPN with DD-WRT

By On June 14, 2011 · In Asterisk, Mobile VoIP, QoS - Quality of Service, SIP protocol, VoIP over VPN

We have two sites with local networks interconnected by VPN tunnel and we have a VoIP server at one of the sites. The server side based on a PC with Windows 2003 Server (also acting as VoIP and VPN server) installed. On the other side we have a DD-WRT as the VPN client.

As we [...]

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Digital Telephony

By On June 13, 2011 · In Asterisk, SIP protocol

Analog telephony is almost dead as you may know. In the PSTN, the famous Last Mile is the final remaining piece of the telephone network still using technology pioneered well over a hundred years ago.

“The Last Mile” is a term that was originally used to describe the only portion [...]

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Analog Telephony

By On June 13, 2011 · In Asterisk, SIP protocol

Analog telephony is almost dead, but you should understand how it works prior you can imagine and get all the advantages on the Digital Telephony.

The purpose of the Public Switched Telephone Network (PSTN)  is to establish and maintain audio connections between two endpoints in order to carry speech. Although [...]

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Mobile VoIP – the future of mobile communications

By On June 13, 2011 · In Android, Asterisk, Mobile VoIP, SIP protocol

 

The vast potential of VoIP for businesses has already showing its signs of gaining importance in corporate market in last decade including “The disruptive” call center and outsourcing who partially or completely relaying on VoIP based services for they service offerings.

The new ” Cool corporate stuff ” for the market is [...]

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Typical VoIP Problems: not right codec, packet loss, jitter, out‐of‐order packets

By On June 9, 2011 · In Asterisk, Mobile VoIP, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality

Due to human perception, VoIP is much more sensitive to certain network conditions that are considered well within spec for most applications.

Network issues such as packet loss, jitter, and packet sequence errors are inherent to IP networks, and are well corrected and tolerated by data transfer protocols. Voice transmissions are real‐time by the nature; [...]

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What is SIP?

By On June 9, 2011 · In Asterisk, SIP protocol, VoIP calls quality

The Basics

The Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, [...]

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QoS and VoIP – how to set priority for VoIP calls with TC

By On June 9, 2011 · In Asterisk, Prioritization and traffic shaping, SIP protocol, VoIP calls quality

The quality of  VoIP phone calls for sure depends on many factors. The main questions that we should understand are: how good is your termination provider and the carrier they are using for the prefix you’ve dialed? What codec is used? Do you have enough bandwidth for real time stream? In most cases even if all [...]

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