Showing posts with label SIP. Show all posts
Showing posts with label SIP. Show all posts

Tuesday, August 16, 2011

PSTN vs VoIP


VoIP (Voice over Internet Protocol) is going to be more popular around the world comparatively with PSTN (Public Switched Telephone Network). The main reason for growing market for VoIP over PSTN is availability, reliability and low cost etc. In this article I will show that there is no reason to limit the expectations to achieve only the same level of quality as in PSTN.

By deploying wideband codec, much better voice quality can be achieved. But using narrowband codec there are some standard solution that can be used in VoIP to get finest quality voice. Such as, traditional PSTN solution does not use the available spectral bandwidth, but these can be done easily in VoIP system. Implementing a wideband codec or expanding the bandwidth of narrowband codec does not automatically guarantee great quality. There are many potential pitfalls when deploying VoIP. There are some issues need to be discussed that effect the final voice quality over PSTN or VoIP. In this article I will discuss what level of quality can be achieved and describe how this can be implemented.



Speech Signals and Speech Coding Sampled digital signals can hold frequency content up to half the sampling frequency. As we know, a young adult has a hearing span from about 20 Hz to 20,000 Hz.
And the sampling frequency of CD audio is chosen to be 44.1 kHz, which is more than double that of the highest frequency perceivable by most humans. Legacy telephony solutions are narrowband, which seriously limits the achievable quality. Wideband codec could potentially be used in digital telephone systems, but this has never been practical enough to gain any real interest

In traditional telephony applications such as PSTN, the speech bandwidth is restricted much more than the inherent limitations of narrowband coding. Typical telephony speech is band limited to 300 Hz to 3400 Hz (listen to Sound Pure digital connections are found in enterprise environments.
Due to poor connections or old wires, significant distortion is often generated in the analog part of the phone connection, a type of distortion that is entirely absent from VoIP implementations. The cordless phones so popular today also generate significant amounts of analog distortion due to radio interference and other implementation issues.



Calls between two parties in a PSTN (Public Switched Telephone Network) are connect by a series of private or public switches. The resulting fixed communications link is dedicated for the duration of the call. When an individual makes a phone call over a circuit-switched network, a connection is made between the providers PBX and the local telephone company, also known as the PSTN. Depending on the destination, this connection might extend to the national or international exchange before reaching another local exchange, where it will be passed on to the PBX and the person who receives the call. This end-to-end link, established by a series of public and private switches, is 100% dedicated on a single, per-call basis and cannot be shared or used for another function as long as the call is in progress. For this reason, these dedicated circuits cannot be shared and the carrier bills the call on a time and distance rate.

But the Internet does not use switches to link calling parties. Instead, the analog voice signal is digitized by an Internet Protocol (IP) and broken up into thousands of small data packets by a router – the VoIP equivalent to a switch. These data packets are sent, or routed, over the public Internet to their destination, enabling calls to bypass the PSTN entirely.


Other article can be read:

Thursday, August 11, 2011

What is H.323 Signaling Protocol - A basic


To provide some special services through the internet (IP based network) such as real-time audio / video communication, data communication H.323 is a standard signaling protocol which components, protocols and procedures. H.323 is part of a family of ITU-T recommendations called H.32x that provides multimedia communication services over a variety of networks.

H.323 was invented to provide various multimedia applications through LAN (Local Area Network). But due to its popularity and rapidly growing services it has evolved in needs of VoIP (Voice over Internet Protocol). One important strength of H.323 was the relatively early availability of a set of standards. It was not only used to define basic call models but in addition to supplementary services that can easily involved in Business Communications. The first VoIP standard protocol is H.323 to adopt IETF standard RTP to transport audio and video communications over the internet (IP network). H.323 standard specifies four kinds of components, when networked together, provide the "point-to-point" and "point-to-multipoint" multimedia-communication services. These are:
  • Terminals
  • Gateway
  • Gatekeepers
  • Multipoint Control Unit

Terminals: Used for real-time bi-directional multimedia communications, an H.323 terminal can either be a personal computer (PC) or a stand-alone device, running an H.323 and the multimedia applications. It supports audio communications and can optionally support video or data communications. Because the basic service provided by an H.323 terminal is audio communications, an H.323 terminal plays a key role in IP–telephony services. The primary goal of H.323 is to inter-work with other multimedia terminals. H.323 terminals are compatible with H.324 terminals on SCN and wireless networks, H.310 terminals on B– ISDN, H.320 terminals on ISDN, H.321 terminals on B–ISDN, and H.322 terminals on guaranteed QoS LANs. H.323 terminals may be used in multipoint conferences.

Gateways: A gateway connects two dissimilar networks. An H.323 gateway provides connectivity between an H.323 network and a non – H.323 network. For example, a gateway can connect and provide communication between an H.323 terminal and SCN networks ( SCN networks include all switched telephony networks, e.g., public switched telephone network [PSTN] ). This connectivity of dissimilar networks is achieved by translating protocols for call setup and release, converting media formats between different networks, and transferring information between the networks connected by the gateway. A gateway is not required, however, for communication between two terminals on an H.323 network.

Gatekeepers: A gatekeeper can be considered the brain of the H.323 network. It is the focal point for all calls within the H.323 network. Although they are not required, gatekeepers provide important services such as addressing, authorization and authentication of terminals and gateways; bandwidth management; accounting; billing; and charging. Gatekeepers may also provide call-routing services.

Multipoint Control Units: MCUs provide support for conferences of three or more H.323 terminals. All terminals participating in the conference establish a connection with the MCU. The MCU manages conference resources, negotiates between terminals for the purpose of determining the audio or video coder/decoder (CODEC) to use, and may handle the media stream. The gatekeepers, gateways, and MCUs are logically separate components of the H.323 standard but can be implemented as a single physical device.


Key Benefits of H.323:

Codec Standards: 
H.323 establishes standards for compression and decompression of audio and video data streams, ensuring that equipment from different vendors will have some area of common support.

Interoperability:
Users want to conference without worrying about compatibility at the receiving point. Besides ensuring that the receiver can decompress the information, H.323 establishes methods for receiving clients to communicate capabilities to the sender. The standard also establishes common call setup and control protocols.

Network Independence:
H.323 is designed to run on top of common network architectures. As network technology evolves, and as bandwidth management techniques improve, H.323-based solutions will be able to take advantage of those enhanced capabilities.

Platform and Application Independence:
H.323 is not tied to any hardware or operating system. H.323-compliant platforms will be available in many sizes and shapes, including video-enabled personal computers, dedicated platforms, IP-enabled telephone handsets, cable TV set-top boxes and turnkey boxes.

Bandwidth Management:
Video and audio traffic is bandwidth-intensive and could clog the corporate network. H.323 addresses this issue by providing bandwidth management. Network managers can limit the number of simultaneous H.323 connections within their network or the amount of bandwidth available to H.323 applications. These limits ensure that critical traffic will not be disrupted.

Flexibility:
An H.323 conference can include endpoints with different capabilities. For example, a terminal with audio-only capabilities can participate in a conference with terminals that have video and/or data capabilities. Furthermore, an H.323 multimedia terminal can share the data portion of a video conference with a T.120 data-only terminal, while sharing voice, video, and data with other H.323 terminals.

Inter-Network Conferencing:
Many users want to conference from a LAN to a remote site. For example, H.323 establishes a means of linking LAN-based desktop systems with ISDN-based group systems. H.323 uses common codec technology from different videoconferencing standards to minimize transcoding delays and to provide optimum performance.

Related Other Topics:

Friday, July 29, 2011

What is SIP Dialer - a brief discussion


A SIP Dialer is called a softphone dialer or Mobile dialer or PC dialer which is used to make phone calls directly to any land phone number or mobile number in the world using VoIP (Voice over Internet Protocol). It is one kind of call center management software which is use in many Call Centers for their regular communication with their client and customers. SIP stands for Session Initiation Protocol.

There are some special advantages of using this kind of call center management software or SIP dialer. This are able to work in both Unicast and Multicast server. Through Unicast the conversation take place between two parties and for Multicast the conversation take place more than two parties which is called teleconferencing. There are some other special features for which the SIP dialer is going to be popular day by day.

SIP dialer or softphone has a number of advantages like:
  • Availability: It is available to download for free on the internet. Many provider provide free version of a SIP dialer to download, user can make calls using this free version directly to any land line or mobile phone number in the world.

  • Cost effectiveness: Using a SIP dialer you can make calls via VoIP (Voice over Internet Protocol), which is the most cost effective way to make phone calls. Traditional phone line or mobile phone can be used to make calls by paying a huge amount of bill.

  • Time saving ability: Some of its special features like teleconferencing facility, broadcasting recorded messages or SMS to a large group of people simultaneously save time when using SIP dialer services.

  • A SIP Dialer is very much predictive, Call centers use this kind of Softphone regularly for their business effectiveness, simple to use functions etc. Using this dialer communication with potential customer and clients is getting more easy and reliable.

  • After downloading the software in the computer it becomes easy to send single SMS or a recorded message to a client or multiple recorded messages or SMS to thousands of recipients.

  • A SIP dialer can easily installed in computer or standard mobile handset. Users can make VoIP phone calls directly from their computer or mobile using SIP dialer. At present people around Middle East countries like KSA, UAE, Qatar, Oman etc. are using this service to make overseas calls. Voice and video calls can also be making from a SIP dialer. The main requirement to make call is internet connection while using in both computer and mobile handset. The quality of voice and video conversation is depend on some other things like voice carrier route, softswitch, byte saver etc. If all these requirements got fulfilled than this kind of calling method is one of the best in the world for cost effectiveness.

Some other use of SIP Dialer:
  • Corporate business offices use SIP dialer service due to its good quality video and voice (single and multiple) services. SIP can sometimes eliminate the need of overnight travel to a business meeting.
  • Doctors who are giving patients prescription around the world using teleconference are using this service regularly.
  • Many university / college / school use this service to taught courses in the classroom or outside the classroom.

Related topics:

Saturday, July 23, 2011

What is SIP - Session Initiation Protocol

To provide advanced telephony service over the Internet (VoIP) a Protocol named Session Initiation Protocol (SIP) has been developed by the IETF (Internet Engineering Task Force). Session Initiation Protocol or SIP is signaling protocol used to established sessions in an IP network like two-way call (IP-2-IP call, Net-2-Phone call etc), a collaborative multi-media conference system. SIP calls may be terminal to terminal, or they may require a server to intercede. If a server is to be involved, it is only required to locate the called party. For inter-working with non-IP networks, Megaco and H.323 are required. Often vendors of VoIP equipment integrate all three protocols on a single platform.

Various telecommunication services like Mobile dialer or PC Dialer service, Calling card services, Click and dial from Web page, Instant messaging, IP Centrex service and many other Voice-enriched E-Commerce services has become possible with the help of Session Initiation Protocol or SIP.

SIP has been designed upon some other protocol like HTTP (Hypertext Transfer protocol), SMTP (Simple Mail Transfer Protocol). SIP borrows most of its syntax and semantics from the familiar HTTP. In an IP-based network it is used to setup, change or end calls between two or more users.

In two method calls can be setup using SIP called, Redirect and Proxy and the server are designed to handle these modes. Both modes issue an “invite” message for another user to participate in a call. The redirect server is used to supply the address (URL) of an unknown called addressee. In this case the “invite” message is sent to the redirect server, which consults the location server for address information. Once this address information is sent to the calling user, a second “invite” message is issued, now with the correct address.

The following features of SIP are playing a major role in the field of VoIP (Voice over Internet Protocol):

Media negotiation: The inbuilt SIP mechanism that allow concession of the media used in a call, enable selection of the proper codec system to making a call between various devices. Thus, simple devices can make VoIP phone call, using the selected codec system.

Feature Negotiation: This feature allows the group involved in a call to agree on the features supported. This may be a multi-party call. All the parties can support the same level of features. For some codec video may or may not be supported; as any form of MIME type is supported by SIP. During making a call one user can bring other users onto the call or cancel the call or may placed or hold the call.

User location and name translation: It ensure that the participation of the second user in the call wherever he is located. Carrying out any mapping of expressive information to location information. Ensuring that details of the nature of the call (Session) are supported.

Changes of Call features: Using SIP a user can setup Voice-Only mode, but during making a call the caller need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call.


Some special features of Session Initiation Protocol:
  • SIP messages are text based and hence are easy to read and debug. Programming new services is easier and more intuitive for designers.
  • SIP re-uses MIME type explanation in the similar way that email clients do, so applications related with sessions can be launched automatically.
  • SIP re-uses a number of existing and mature internet services and protocols such as RTP, DNS, RSVP etc. No new services have to be introduced to support the SIP infrastructure, as much of it is already in place or available off the shelf.
  • SIP extensions are defined clearly, enabling VoIP service providers to add them for new applications without damaging their own networks. Older SIP-based equipment in the network will not hamper newer SIP-based services. An older SIP implementation that does not support technique / title utilized by a newer SIP application would simply ignore it. SIP is transport layer independent. Therefore, the underlying transport could be IP over ATM.
  • SIP uses UDP - User Datagram Protocol, as well as the TCP - Transmission Control Protocol, lithely connecting users independent of the primary communications.
  • SIP supports multi-device feature leveling and negotiation. If a service or session initiates video and voice, voice can still be transmitted to non-video enabled devices, or other device features can be used such as one way video streaming.

SIP sessions use up to four major components:
  • SIP User Agents
  • SIP Registrar Servers
  • SIP Proxy Servers and
  • SIP Redirect Servers.

Together, these systems deliver messages embedded with the SDP protocol defining their content and characteristics to complete a SIP session.

Further reading about SIP.


Other topic:

Wednesday, June 1, 2011

Setting for GRAND STREAM SIP Devices with VoIP Buster

There are various SIP devices in the market GRAND STREAM is a very much well known. In this post SIP settings for this device will be described. SIP settings with VoIP Buster for the following devices:
 
GRANDSTREAM HT286
GRANDSTREAM HT386
GRANDSTREAM HT486
GRANDSTREAM HT496

1. GRAND STREAM HT286:
  • SIP server: sip.voipbuster.com
  • Outbound proxy: sip.voipbuster.com
  • SIP User ID: Your VoIP Buster username, this is the username you used for the registration of your Voip Buster-account
  • Authenticate Password: Your VoIP Buster password, this is the password you used for the registration of your Voip Buster-account
  • Name: Your name, this name will be displayed in the phone screen of the person you are calling
  • Don't change the Advanced Options.

2. GRANDSTREAM HT386:
  • SIP Server: sip.voipbuster.com
  • Outbound Proxy: sip.voipbuster.com
  • SIP User ID: Your VoIP Buster username, this is the username you used for the registration of your Voip Buster-account
  • Authenticate Password: Your VoIP Buster password, this is the password you used for the registration of your Voip Buster-account
  • Name: Your name, this name will be displayed in the phone screen of the person you are calling
  • Local SIP port: 5060
  • Don't change the other options

3. GRANDSTREAM HT486:
 
  • SIP Server: sip.voipbuster.com
  • Outbound Proxy: sip.voipbuster.com
  • SIP User ID: Your VoIP Buster username, this is the username you used for the registration of your Voip Buster-account
  • Authenticate Password: Your password, this is the password you used for the registration of your Voip Buster-account
  • Name: Your name, this name will be displayed in the phone screen of the person you are calling
  • Don't change the Advanced Options

4. GRANDSTREAM HT496:
  • SIP server: sip.voipbuster.com
  • Outbound Proxy: sip.voipbuster.com
  • SIP User ID: Your VoIP Buster username, this is the username you used for the registration of your Voip Buster-account
  • Authenticate Password: Your VoIP Buster password, this is the password you used for the registration of your Voip Buster-account
  • Name: Your name, this name will be displayed in the phone screen of the person you are calling
  • Local SIP Port: 5060
  • Don't change the other options

Before configure any kind of SIP devices you must read the User manual carefully provided with the devices. For more details go to voipbuster.com.

Saturday, May 28, 2011

SIP Settings for LINKSYS Devices with VoIP Buster

A little discussion about how to configure SIP settings for the following LINKSYS Devices with VoIP Buster:

LINKSYS WIP 330
LINKSYS SPA 3102


1. SIP Settings for LINKSYS WIP 330:


  • SIP Proxy: voipbuster.com
  • SIP Proxy Port: Port number 5060
  • Registrar Address: sip.voipbuster.com
  • Registrar Port: Port number 5060
  • Expiry Time (sec): 3600 seconds
  • Display name: You can fill-in your name, this name will be displayed in the phone screen of the person you are calling
  • Registrar Name: Your username, this is the username you used for the registration of your Voip Buster-account
  • Registrar Password: Your password, this is the password you used for the registration of your Voip Buster-account



2. SIP Setting for LINKSYS SPA 3102:


  • Line Enable: Yes
  • SIP Port: Port number 5060
  • Proxy: sip.voipbuster.com
  • Register: 'Yes'
  • Make Call Without Reg: 'Yes'
  • Register Expires: 3600 seconds
  • Ans Call Without Reg: 'No'
  • Display name: You can fill-in your name, this name will be displayed in the phone screen of the person you are calling
  • User ID: Your username, this is the username you used for the registration of your Voip Buster-account
  • Password: Your password, this is the password you used for the registration of your Voip Buster-account
  • Use Auth ID: 'No'

Besides this setting please all are requested to read the user manual provide with the SIP Devices.

Further Reading:

Wednesday, May 25, 2011

SIP Settings for VoIP Buster with AVM Fritz box Devices

In this post VoIP Buster SIP Settings for following devices will be described:

AVM Fritz! box FON 7050
AVM Fritz!box FON ATA 1020

1. AVM Fritz! box FON 7050: See the screen shot and fill up the following information in the provided fields.




  • Internet number: your VoipBuster username (if available)
  • User Name: fill in your Username, this is the username you used for the registration of your VoIP Buster-account
  • Password: fill in your Password, this is the password you used for the registration of your VoIP Buster-account
  • Password confirmation: fill in your password again
  • Registrar: this is sip.voipbuster.com
  • Proxy server: this is sip.voipbuster.com
  • Overview - number 1, 2 or 3: your VOIP phone number (if available)


2. AVM Fritz! box FON ATA 1020: See the screen shot and fill up the following information in the provided fields.




  • Internet number: you can fill-in e.g. your name or telephone number
  • User name: fill in your Username, this is the username you used for the registration of your VoIP Buster-account
  • Password: fill in your Password, this is the password you used for the registration of your VoIP Buster-account
  • Password confirmation: fill in your password again
  • Registrar: this is sip.voipbuster.com
  • STUN-Server: this is stun.voipbuster.com

Please besides these setting read your user manual of the SIP devices. For more information please visit VoIP Buster web site: www.voipbuster.com

Other related post:

Sunday, January 30, 2011

Limits of Traditional Telephony System over VoIP


In this post I am going to discuss about some limits of traditional Telephone (PSTN) system comparing with VoIP system. The PSTN's capabilities are largely proportional to its physical connections, because every call must have a circuit, or loop, set up at the beginning of the call and torn down at the end. The PSTN telephone system can does this calling easily but there are some limitations associated with its circuit-switched nature. The whole system took lots of year to complete. And many PSTN companies are still working on it to make this service more popular to users. Still some parts of this PSTN does not support Caller ID.

Capacity limits is another disadvantage of this kind of telephone system. The reliability of a call’s sound is reproduction is limited to the available Bandwidth between the recipient and caller. And the number of calls between two or more offices depends on the availability of Physical Circuits that are exists between them.
Another main problem of traditional telephone system is the cost. The telephone companies and phone equipments manufacturers are hardly trying to reduce the call cost and capacity limitations. PBX feature like LCR – Least Cost Routing and High-density digital circuits such as T1, T3 has brought the cost of high density telephony down. Recently long distance calling rate is going to be cheaper and the cost of on-premises PBX hardware and feature – rich business telephone has dropped over time.
Since the telephone system features were considered as a competitive advantage in personal and business use, and users began searching for a low cost telephone system. But the question was how ?

Enterprise telephony innovators began looking on the Internet for the answer how to minimize the cost. Because of core differences in engineering philosophy and many years of additional discourse on the matter, the Internet is superior in many ways to traditional voice networks.

Communication Protocol on the internet are in a constant state of improvement. So using this technology excellent voice can be generated using bandwidth which will be more efficient. On the internet the capacity of calls can handle through software instead of Hardware like PSTN, PBX etc. IP Networks grow in capacity as software improves. The software on the IP telephony system use low cost PCs, its another advantage. Hardware up gradation is not too much expensive. Upgrading software can increase the capacity and it is easier to scale the capacity on IP networks rather than circuit based telephone system.

While the PSTN is quite reliable, it is far less disaster-proof than IP networks. The Internet Protocol permits redundancy and fail over capabilities that are inexpensive and relatively easy to implement and maintain. Geographic diversity, a technique used on data networks to circumvent local connectivity interruptions, is very easy for the enterprise to achieve with the Internet, but more difficult on the PSTN. For example, you can connect to two Internet service providers and use the same set of IP addresses with both, thanks to the BGP standard, but it's nearly impossible to use the same set of phone numbers with two telephone companies. Border Gateway Protocol, RFC 1105.

Because most modern enterprise networks use the same protocols as the Internet, it was only a matter of time before the advantages of those protocols began appealing to designers of voice networks. The result of that appeal is an immense technology family called Voice over Internet Protocol, or VoIP.

VoIP is loosely defined as using the TCP/IP Protocol suite to transmit voice conversations, but it is really much more than voice conversation. It can be used to replace traditional telephony in the enterprise or in the home or merely to add some extra features to a traditional telephony system. VoIP can also solve connectivity challenges, like linking traditional PBXs at remote sites together, linking private telephone extensions at a single site together like a PBX, or simply aggregating calls among a few analog phones like a key system.
VoIP can be used to facilitate voice communications on many different application substrates, too. It can provide on-demand voice-calling capabilities to users of a customer service web page and allow people to use their personal computers as fully featured business telephones. It can play a role in bridging cell phone and landline systems, too.

VoIP technologies can even run the entire enterprise voice telephone network. The VoIP technology family can even facilitate video streaming, conferencing, whiteboard applications, and instant text messaging, challenging the traditional distinction between data and voice networks.

Of course, VoIP needs a TCP/IP network in order to operate. Until recently, TCP/IP networks were less abundant than were connections to the PSTN. And even when TCP/IP began to rise in popularity, many private networks still weren't connected to the Internet, so the PSTN was always more appealing than the Internet for voice applications. But today, all of that has changed. There are an estimated 36 million permanent, private TCP/IP networks in the United States, and about 30 million of them are linked via high-speed connections to the Internet

When data networks like the Internet are used to carry real-time media traffic, they are called converged networks. The process of implementing real-time media applications, such as IP telephony, on a data network is called convergence. Networks (such as the Internet and software-based call management). VoIP's biggest advantages over traditional telephony are scalability and infrastructural cost savings, though easier integration between telephony and computer applications is a big attraction, too.

Enterprise implementers considering VoIP are comparing scalability against that of traditional PBX equipment and discovering that VoIP's basis in software gives it a big advantage. While a small PBX built for 100 telephones is quite costly to scale up to 500 phones, for example, a VoIP call-management solution can usually scale up just as servers and network hardware doand 500 phones is not a particularly tall order for even a moderately equipped VoIP setup.


Reasons to Choose Voice over IP
  • VoIP devices are easier and cheaper to maintain because they leverage the cor-porate data network rather than their own single-purpose voice-only network.
  • VoIP increases the value of the Internet by using it for voice communications.
  • Integrating telephony with computer applications is much easier with VoIP than with traditional voice systems, because in VoIP settings, call management tends to be more open, standardized, and software-driven.
  • VoIP can scale more cheaply than traditional voice systems because it often uses off-the-shelf, frequently updated PC hardware.
  • It allows for more centralized administrative control than a traditional PBX.
  • Managing a VoIP network is accomplished using the same network that carries the voice information itself, unlike the PSTN with its SS7 protocol.
  • Survivability against disasters and network outages is easier to achieve with TCP/IP than with traditional voice systems, due to its basis in software and TCP/IP's remarkable resilience measures and routing protocols.
  • Much of your existing phone equipment can interface with VoIP systems using analog terminal adapters, or ATAs.
  • VoIP's acronym is fun to say ("voyp").

Cost savings are attracting businesses to VoIP. Since VoIP runs over a data network, it has the same facilities requirements as a data network. Whereas traditional telephony devices such as a PBX normally require separate local area wiring for analog and digitial phone connections, this wiring usually can't be used for local area data networking because it doesn't provide ample resistance to interference and attenuation. With VoIP, the same wiring is used for both data and voice, since the voice is carried within the data network.
The cost of future expansion is almost always less expensive with VoIP than with traditional telephony. Because it can be centrally administered more easily than traditional systems, VoIP allows system expansion, ongoing security enforcement, and back-office call accounting to be cheaper over time.


Other related posts:

Saturday, October 16, 2010

Cheap and Free phone calls using VoIPCheap


Make Free Phone calls over the internet from your computer or mobile using VoIPCheap. To make a free call, just install the VoIPCheap application in your computer, connect your Headset to PC, get registered, dial you destination number and start talking. The application is absolutely free to download.

You can make calls for free to other VoipCheap user through the internet. See more about minimum requirements of the system. Or if you want to use this application from your mobile just click HERE.


You can make free phone calls to the following destinations:
  • Andorra
  • Argentina
  • Australia
  • Austria
  • Belgium
  • Canada
  • Czech Republic
  • Denmark
  • Estonia
  • France
  • Germany
  • Hong Kong (+mobile)
  • Hungary
  • Iceland
  • Ireland
  • Italy
  • Luxembourg
  • Malaysia
  • Monaco
  • Netherlands
  • New Zealand
  • Norway
  • Poland
  • Portugal
  • Russian Federation [moscow]
  • Russian Federation [st Petersburg]
  • South Korea
  • Spain
  • Sweden
  • Switzerland
  • United Kingdom
  • United States (+mobile)

With VoIPCheap you can make international phone calls and send SMS to A to Z destination with a low and reliable rate. Visit Site for detail rate plan.

You can buy credit at any time. VoIPCheap support lots of online payment method. You can select any method which suits you best. Like Visa, Visa Electron, Master Card, PayPal, MoneyBookers etc.





To make your first call using VoIPCheap you need to get registered. Username and password is required in order to use VoIPCheap. After getting registered follow some test related to sound quality. The application will ask you to check your internet bandwidth and hardware specification.



You can use your own SIP devices and for those who use at least 5000 minute monthly can get the opportunity to open a wholesale account with a low rate to A-Z destinations.

Start making you phone calls now and enjoy.


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