voip-sip.org - Internet VoIP calls, International SIP termination, VoIP hardware
  • 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
    • VoIP Codecs
    • Cisco ATA186 + Asterisk Config
  • Who we are?
  • How to start
  • Free SIP account
  • Configs
    • Grandstream Budgetone
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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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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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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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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VoIP Quick start guide

By Nik On November 9, 2011 · In Open source, QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over VPN

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 [...]

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Voip via VPN SSL voice quality

Running VoIP via VPN (SSL) – voice quality

By Nik On October 18, 2011 · In QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over SSL VPN, VoIP over VPN, VoIP via VPN

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 [...]

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Skype App directory

By Nik On August 31, 2011 · In VoIP calls quality

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 [...]

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

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

By Nik On August 23, 2011 · In Mobile VoIP, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over VPN

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 [...]

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Skype hack - free calls

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

By Nik On August 10, 2011 · In VoIP calls quality

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 [...]

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Nimbuzz growing even without Skype

By Nik 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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Skype protocol hack

By Nik On June 24, 2011 · In SIP protocol, VoIP calls quality

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. [...]

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Call DSN number

By Nik On June 23, 2011 · In VoIP calls quality

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 [...]

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How to make your VoIP calls private and confidential

By Nik 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 Nik 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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VoIP Codec: Payload size

By Nik 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 Nik 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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The QoS Dilemma

By Nik On June 15, 2011 · In Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality

The problem with IP is that, like Ethernet, it is a connectionless technology and does not guarantee bandwidth.  Specifically, the protocol will not, in itself, differentiate network traffic based on the type of flow to ensure that the proper amount of bandwidth and prioritization level are defined for a particular type of application. By contrast, [...]

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VoIP calls over satellite links

By Nik 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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frantic-phone-call

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

By Nik 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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SIP based VoIP service is on path to become one of the most significant protocols since HTTP and SMTP

By Nik On June 9, 2011 · In QoS - Quality of Service, SIP protocol, Uncategorized, VoIP calls quality

Understanting SIP

The growing thirst among communications providers,their partners and subscribers for a new generation of IPbased services is now being quenched by SIP – the SessionInitiation Protocol.  An idea born in a computer sciencelaboratory less than a decade ago, SIP is the first protocolto enable multi-user sessions regardless of media contentand is [...]

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What is SIP?

By Nik 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 Nik 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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  • 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
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    • 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
    • Sip on Android
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  • More to read on VoIP

    • About
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    • How to start with VoIP telephony
    • Multiplexing RTP Data and Control Packets on a Single Port
    • On-line payments
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    • What is VoIP and what it can do for you
    • Why should you switch to VoIP services?
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  • Plain explain: IP-phone
    We all strive to get a quality telephony, but when we […]

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