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
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 [...]

Continue Reading →
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.

Continue Reading →
field_telephone_exchange_wwi

History: Kellogg Field Phone (World War I)

By Nik On October 19, 2011 · In History, Mobile VoIP, SMS to email

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 [...]

Continue Reading →

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 [...]

Continue Reading →

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 [...]

Continue Reading →

Microsoft enabling the ability to eavesdrop on VoIP conversations

By Nik 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 [...]

Continue Reading →

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 [...]

Continue Reading →

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 [...]

Continue Reading →

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 [...]

Continue Reading →

Asterisk and Google Voice

By Nik 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 [...]

Continue Reading →

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 [...]

Continue Reading →

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 [...]

Continue Reading →

VoIP client behind a VPN with DD-WRT

By Nik 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 [...]

Continue Reading →

Mobile VoIP – the future of mobile communications

By Nik 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 [...]

Continue Reading →
sipdroid_settings_sip_calls_android_screenshoot_3

Sip on Android

By Nik On June 10, 2011 · In Android, Mobile VoIP, SIP protocol

There is a very interesting project started recently, called Sipdroid – VoIP software for Google Android operating system using SIP (Session Initiation Protocol) and it’s actually Java SIP stack contributed by MJSip.

Sipdroid is open source free software released under the GNU General [...]

Continue Reading →
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; [...]

Continue Reading →
  • 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
    • Sip on Android
  • VoIP SIP IP telephony tags

    android Asterisk bandwidth bandwidth requirements best effort cellular codec delay encryption options facebook free calls g273 g726 g729 google gsm high latency How it works ilbc IPsec issues jitter listen to voip mobile Nimbuzz nokia order packet payload. g711 privacy protocol analyzer pstn QoS - Quality of Service QoS protocols real-time applications record voip calls satellite link sipdroid SIP protocol Skype speex TLS voice quality voip voip becomes social
  • Quick navigation

    • Android (6)
    • Apple (1)
    • Asterisk (22)
    • Cloud VoIP (1)
    • FaceTime (1)
    • Google Voice (6)
    • History (2)
    • IT news (2)
    • Mobile VoIP (16)
      • Symbian (1)
    • Non VoIP news (2)
    • Open source (10)
    • Prioritization and traffic shaping (9)
    • QoS – Quality of Service (11)
    • SIP protocol (32)
      • Cisco (2)
    • SIP termination (2)
    • SMS to email (1)
    • Softphone VoIP (1)
    • Ubuntu (1)
    • Uncategorized (1)
    • VoIP calls quality (24)
    • VoIP industry news (4)
    • VoIP over SSL VPN (1)
    • VoIP over VPN (8)
    • VoIP service (2)
    • VoIP via VPN (1)
  • More to read on VoIP

    • About
    • About Mark Spencer
    • Asterisk SIP Media NAT
    • Browser-based VoIP: web page code to call over IP (to your VoIP account)
    • Choosing the right provider
    • Cisco ATA186 notes
    • Free sip account
    • Grandstream Budgetone configuration manual
    • How to start with VoIP telephony
    • Multiplexing RTP Data and Control Packets on a Single Port
    • On-line payments
    • VoIP Codecs
    • VPN: UDP or TCP?
    • What is VoIP and what it can do for you
    • Why should you switch to VoIP services?
  • Blogroll

    • Asterisk™: The Definitive Guide (new window) “Asterisk has been emblematic of the way that open source software has changed business—and changed the world”
    • Blog Jon FreeSWITCH VOIP SIP Asterisk Linux Open Source
    • Business.com Business.com is one of the Web’s largest directories for business products and services
    • Ubuntu how-to www.ubuntuka.com Miscellaneous Ubuntu Tips, Tricks and Hints
  • Our Twitter – latest

    • Introduction to Voice over IP (VoIP) - One important step into adopting VoIP is... voip-sip.org/introduction-t… 5 months ago
    • How to configure Internet tel. and SIP... voip-sip.org/how-to-configu… 5 months ago
    • Hey! - To all people who has an experience with Asterisk and Linux - quote of the day: "I have not failed.... voip-sip.org/hey/ 5 months ago
    • What is VoIP termination - Many people keep asking me - what is VoIP Termination?... voip-sip.org/what-is-voip-t… 5 months ago
    • How to configure Internet tel. and SIP... voip-sip.org/how-to-configu… 5 months ago
    Follow @sipcalls
  • Protected by Copyscape DMCA Copyright Detector
"Introduction to Voice over IP (VoIP) - One important step into adopting VoIP is... http://t.co/fYdnQ7Jc" — sipcalls

voip-sip.org – Internet VoIP calls, International SIP termination, VoIP hardware

Pages

  • About VoIP
  • Asterisk IP-PBX
  • Who we are?
  • How to start
  • Free SIP account
  • Configs

The Latest

  • Plain explain: IP-phone
    We all strive to get a quality telephony, but when we […]

More

Thanks for dropping by! Feel free to join the discussion by leaving comments, and stay updated by following ourVoIP-SIP.org Twitter.
© 2013 voip-sip.org
Platform by PageLines