IP Phones

An IP phone is a device which uses an internet connection to make and receive phone calls. Essentially, there are two types of IP phones: Hardware based IP phones that use an adaptor to plug your standard home telephone into your home network and software based IP phones (or soft phones) that consist of software which allows telephone calls to be made directly from your computer.

Hardware-based IP Phones

Hardware based phones consist of an ATA (Analogue Telephone Adaptor) which connects your PSTN to your home network. Popular ATA products in the market include CISCO’s ATA 180 Series and the Sipura SPA-3000. Calls are made directly using the telephone plugged into the network.

PSTN (Public Switched Telephone Network) refers to the collection of interconnected telephone networks that exist in today’s world. It is also popularly referred to as POTS (Plain Old Telephone Service). When you make a call, your local carrier routes it through a series of physical switches which culminate in your phone being connected to the other person’s.

Software-based IP phones

Software based phones simply refer to pieces of software that run on your computer and allow you to make telephone calls through them. Popular products include software such as Skype, Google Talk and the Yahoo Messenger. The latter two allow you to make phone calls only to people using the same service while Skype transcends these boundaries and allows you to call regular phone users (at a cost of course). Soft phones need to be used with peripheral speakers and microphones.

Advantages

IP phone services are very cheap compared to traditional phone services. Their biggest savings come from the lack of taxes and fees. One of the key advantages of IP phones is their ability to maintain local long distance. IP phones work with virtual telephone numbers. By using a virtual phone number with an area code similar to the place you are calling to, local calls can be made from even half way across the world. Similarly, anyone contacting you from that area will also be making a local call thus avoiding any tolls or cess in the process.

IP phones also help integrate various tasks such as managing caller id’s, call waiting and forwarding etc from the comfort of a personal computer.

Disadvantages

On the flip side, however, IP phone services are not always reliable. Voice quality generally depends on the quality of your internet connection and 911 phone calls cannot always be made using these phones. The ATA (Analogue Telephone Adaptor) also runs on electric power and stops working in case of a power failure or malfunctioning internet connection. High end ATA’s however automatically switch to the POTS line when this happens.

The PSTN

The term PSTN (Public Switched Telephone Network) refers to the collection of interconnected telephone networks that exist in today’s world. It is also popularly referred to as POTS (Plain Old Telephone Service). When you make a call, your local carrier routes it through a series of physical switches which culminate in your phone being connected to the other person’s. This entire process is referred to as circuit switching.
The PSTN is also the corner stone of the internet’s infrastructure. Since ISP’s (Internet Service Providers) pay long distance providers for access to their infrastructure and share circuits through packet switching, Internet users avoid having to pay usage fees to anyone other than their ISP’s.

Top Reasons for Using VoIP

The number one reason to switch to VoIP technology for telephone service is cost reduction. From that base, VoIP is able to provide some compelling features which makes switching even more attractive.

Eliminating Phone Lines

With VoIP service, you can cancel your traditional phone service through your local telephone company and place all of your telephone calls over your broadband Internet connection.
For a residential customer, this will save around $40 a month. For business customers, the savings can be thousands of dollars a month.

Eliminating Long Distance Charges

VoIP technology can also save money on long-distance charges. Most residential and business telephone customers pay per-minute fees for long-disatance telephone calls. VoIP can reduce or eliminate those long-distance fees.
This saving is especially valuable with International calls, where per-minute charges for traditional telephone calls can be very expensive.

Number Portability

With VoIP service, you can take your phone number anywhere you go, easily. If you have a Chicago number and you move to New York, you can keep your Chicago number. This is very convenient for friends and family to keep in contact with you wherever you go.

Computer Telephony Integration (CTI)

VoIP service providers are designing and implementing new features which implement Computer Telephony Integration (CTI).
For example, VoIP customers may be able to receive their voice messages in e-mail as .WAV file attachments. This can make managing voice mail messages much easier and more powerful, because it enables recipients to archive voicemails or forward them to anyone with an email address.

SPIT

VoIP spam or Spam over Internet Telephony (SPIT) is one of the foreseen future forms of spamming that Internet authorities are preparing for today. With the increasing use and dependence on the Internet for communications and data transfer, malicious software programmers have taken advantage by creating VoIP bots with the ability to harvest data and advertise massively at a very small cost. These advertising methods include email spams, SPIMS or spams over instant messaging applications, malicious bots that generate pop up ads, initiate redirects, etc.
With the inevitable popularity of VoIP over the traditional telephone, authorities are convinced that this is where the next form of spam will come from. In this case, the unsolicited emails will be replaced by video or audio recordings advertising dubious products and services. Prank callers will also take advantage of this new frontier as the new technology becomes more available. This is even more profitable for such users as they can send automated or pre-recorded advertising messages to thousands of users with just one click, making it a very cheap operation to run.

SPIT will also have more impact on users than unsolicited instant messaging and email spam as it has the potential of clogging up the network. Given enough SPIT volume, users may not have any other options that to hang the VoIP phone 'off the hook'.
Other threats include spammers who might take temporary control of a user's systems to launch VoIP attacks on other networks, hackers that will inject profane words in conversations, fake voice mails and viruses that have the ability to use critical bandwidth.
Furthermore, These VoIP bots have the capability of launching automated DDoS or distributed denial of service attacks against rival corporations or users using VoIP with SIP protocols and vulnerabilities. Botnets armed with VoIP-directed software will play a big role in launching these kinds of attacks.

VoIP Built-In Security

VoIP, however, will have the usual array of spam defenses that other forms of Internet communication applications like emails and instant messengers have to combat unsolicited video/voice communication. This will include the stealth mode of instant messenger applications, privacy options as well as spam reporting options.
Other security measures may also include separating the voice and data streams so that in the event that the voice lines do get clogged with traffic, the website traffic will not be affected and will remain operational. Anti-spyware and anti-virus systems coupled with SIP encryption systems designed for VoIP will also help a lot in screening incoming calls and data and detect any instructions in the system. This will prevent DDoS attacks from being launched against your company. Moreover, increased collaboration between ISPs and Internet authorities will also be effective in determining the locations of these spammers as well as blocking calls and data from dubious IP addresses.
Improvements and updates on the security systems for VoIP systems in its initial stages will also play a crucial role in making this communication option a cost effective and reliable alternative than telephones.

RSVP

RSVP (Resource ReSerVation Protocol) is a protocol used in VoIP to manage QoS (Quality of Service).
RSVP works by requesting that required bandwidth and latency be "reserved" for the VoIP telephone call by every network device between the two endpoints.

RSVP is defined in RFC 2205: Resource ReSerVation Protocol (RSVP).
RSVP is a unicast and multicast signaling protocol, designed to install and maintain reservation state information at each router along the path of a stream of data.
The RSVP protocol is used by a host to request specific qualities of service from the network for particular application data streams or flows. RSVP is also used by routers to deliver quality-of-service (QoS) requests to all nodes along the path(s) of the flows and to establish and maintain state to provide the requested service. RSVP requests will generally result in resources being reserved in each node along the data path.

Where you can get VoIP Training

Voice over Internet Protocol (VoIP) is a relatively new technology, because of which individual businesses will need someone who is well versed in the intricacies of VoIP. Companies are often willing to invest in training their IT administrators in VoIP because of the sure returns. But, where do you go to obtain a training course that delineates all the necessary concepts and details you need?
There are various options available for those wanting to get themselves trained in VoIP technology. The primary means of training are through self-study coupled with practical application, classroom training and e-learning. Books, websites and demonstration videos/DVDs can be used as training materials.

Candidates for VoIP Training

VoIP is a combination of telephone connectivity and networking, and because the technology requires expertise in networking and telecommunication, both communication and networking engineers will be the ideal candidates to participate in training in VoIP concepts. Instructors of VoIP generally recommend sending such engineers for training.

VoIP Training Options/Resources

There are basically two types of VoIP training courses – vendor specific and non-vendor specific.

Vendor Specific VoIP training

Vendor specific training is provided in the form of certification courses by leading players in the networking equipment market, like Cisco and Avaya.

Non-vendor Specific VoIP training

The options below should be of great help for non-vendor specific VoIP training.

1. Teracom Training Institute
2. Training City
3. PTT (e-Learning)
4. Global Knowledge
5. Telecommunications Research Associates

The above courses will not lead a participant to a product-specific certification, but will have all the necessary ingredients of VoIP technology and its implementation.

Off-Site vs. On-Site Instruction

Most reputable firms prefer on-site instruction, especially if the office has enough employees to warrant a localized training course. As an alternative, they may invite the trainees to a centralized training facility or offer interactive webinars. The former option is better if you are looking for a comprehensive on-the-job training course, and the latter is appropriate if you want a more subject oriented session on a specific VoIP topic.

VoIP Security

Any technology that involves transfer of data or information is prone to compromised security. It happens with telephones, cell phones, email and Internet transactions. Because VoIP (Voice Over Internet Protocol) has the internet as its mode of transference it's possible to have your Internet-based called intercepted. To make matters worse, there are techno-troublemakers who are armed with the hacking skills needed to eavesdrop on virtually any call over the Internet they want to. It is impossible to ensure total security on information flow over the web including Internet based phone calls. As new technologies emerge with more highly developed security protocols, there will be those who consider it a unique challenge to crack these online defenses rendering security advances antiquated. The Internet has been notorious for alternating security breaches and accompanying fixes.
As VoIP becomes more popular, VoIP security continues to be stressed as a key to advancement of this technology, especially since it will thrive in the realm of the World Wide Web. There are, however, advances in VoIP security that have been utilized by VoIP providers in order to ensure protection of customer's personal information.

VoIP Security is IP Security

VoIP is vulnerable to all security issues that generally affect the traditional IP data networks. This includes viruses, worms and denial of service (DoS), spoofing, port scanning, unauthorized access from a third party. and toll fraud. In short, the same issues you deal with in compromised Internet function can be linked to the use of VoIP technology.

VoIP's Defensive Linemen

The two primary methods of security for VoIP users are tunneling and encryption. These security measures assist in providing a mechanism of trust in the safe use of the VoIP user's personal data. Most VoIP providers use Layer 2 tunneling and an encryption method called Secure Sockets Layer or SSL to keep hackers at bay. Large corporate enterprises are using similar security mechanisms based on encryption for all internal traffic flowing over the VoIP system as well. It is advisable to route all inbound VoIP traffic that flows via a firewall through a proxy server, thus eliminating any direct connection with the internet.
On a larger level, organizations that are using VoIP as a popular mode of communication rely on a multiple level defense that addresses most VoIP security issues. In this scenario, the VoIP network is divided into secure zones protected by layers of firewall, intrusion prevention, and various additional security mechanisms. The advantage with this strategy is that it allows an organization to logically split and secure separate voice and data networks in front of individual voice and data components and between interactive points within the network. A system (like the one just described) should be complete with authentication, controls access (passwords and firewalls), encryption, an audit trail of calls, and facilities. Recording these issues can prevent security issue to a large degree because they are traceable.

Securing Your VoIP Network

While VoIP being internet-based is a key vulnerability, it also has its beneficial side. The years of experience in fending off or foiling internet attacks is experience that can be used in blocking VoIP assaults; the lessons learned in the data networking field are just as applicable to VoIP networking.
One approach that should be given serious consideration is setting up a separate network for VoIP applications, running in parallel but separate from the data network. This may be considered a serious expense item that is incompatible with the perceived savings from VoIP. On the other hand, one has to consider the potential costs involved if both networks become corrupted or disrupted from an attack on one which also disrupts the other.
Here are some other methods for securing a VoIP network:

• – Enable as many of the manufacturer's security protocols as possible, adapting or 'tweaking' these to your own specifications rather than simply following manufacturer's defaults. Keep in mind that hackers and other attackers would probably know these defaults as well.
• – Apply a strong authentication and encryption for both data and voice networks. As noted above, use the lessons learned in dealing with data network security problems to establish a preemptive stance in dealing with potential VoIP security concerns.
• – Work out access controls and authentication protocols to ensure that only legitimate users can gain access to the VoIP network.
• – Use gateway and host-based anti-virus as well as anti-spyware programs to protect crucial VoIP servers. At the same time, consider establishing perimeter security protocols to protect both networks.

A key point to remember is that VoIP is built on already established equipment and applications. The experiences and lessons gained from dealing with security threats to the data network can and should be used in developing security for the voice network.

Conclusion

Because VoIP is a newer technology there is a lot of discussion about its security and reliability. But it may be interesting to note that VoIP is actually more secure than normal email or even bill paying online. You may not need to be too worried about the security issues related to VoIP technology. Many newer technologies are emerging and, given the current trend, it won't take long before VoIP will be as secure as any other communication technology available today. Until then, if you are not sending highly sensitive information over the internet, VoIP is a relatively safe, reliable, and cost effective means of communication.

Videos on VoIP Security

Keeing your VoIP Secure: An Introduction to Cain, ARP, and MITM Attacks.

http://www.youtube.com/watch?v=qt3LaZhGRoQ&feature=player_embedded

How to Compare VoIP Providers

VoIP (Voice over Internet Protocol) is changing the way people communicate. VoIP utilizes a broadband internet connection for routing telephone calls, as opposed to conventional switching methods, providing efficient use of existing Internet connections as well as lowering overall costs. Interestingly, there is no need for any large scale infrastructures; just combine a conventional phone with a broadband Internet connection to utilize a single service with minimal software and hardware support.
VoIP service providers are touting unlimited local and long distance calling for as little as $199 per year. This provides customers with substantial annual savings. There are several VoIP providers offering VoIP service for both residential customers as well as business. However, from a customer's standpoint it is an ideal option to compare several VoIP providers in selecting the best deal.

VoIP Product Features
There are several VoIP providers who claim outstanding services and comprehensive features. Don't be fooled - not all VoIP services are created equal. The VoIP package includes many features that may not be available on traditional phones. The most common VoIP features include 3-way calling and call waiting. As the competition between VoIP providers escalates, some providers are offering additional features to establish branding of their business while attracting additional customers. That's why it's always a good option to compare several VoIP providers to discover the VoIP product features you will get when taking a connection from the provider.

Monthly Rates

One of the main advantages of VoIP is reduced long distance cost and inexpensive local phone service with several enhanced features conventional telephone services are ill equipped to provide. Compare various VoIP providers to know the monthly rates they charge for their service. Selecting an ideal VoIP provider will help you to save up to 75% on expected annual charges.

Using VoIP for International Calling

If you make a lot of international calls, do a bit of research to find a VoIP provider who offers outstanding international services at the best rates. International rates differ from one VoIP provider to another. There are also some carriers which offer unlimited overseas calling. Though this offer is limited to certain countries, check whether the country to which you call falls in this category.

911 Service

Today, majority of the VoIP providers offer E911 service. While selecting a VoIP provider, make sure the provider offers 911 service.

Keeping Your Number

There are many VoIP providers who allow the customers to transfer (port) their current phone number to the VoIP service. Not all VoIP providers offer this service. If you need to change your phone number in this way, then you need to do research on the various VoIP providers to discover whether they offer such services. However, before asking your VoIP provider to switch your current number to the VoIP service, it is advisable to try out the provider's service and make sure that you are satisfied with the end result.

Money Back Guarantee

As VoIP is a relatively new product, most of the VoIP providers will offer a free money back guarantee. As a customer you will be in a risk-free position if your VoIP provider is offers a money back guarantees for up to 30 days.
Comparing various VoIP providers will help you to select the one VoIP service provider whose terms and conditions meet your specific needs and calling pattern, especially if you make regular long distance or international calls.

How to Choose a VoIP phone

The first choice is determining if you want a hardware VoIP phone or a software VoIP phone.
Hardware phones are generally easier to use and do not require a PC. Software phones are usually less expensive and may offer better options for CTI (Computer Telephony Integration).

Choosing a VoIP Phone

With either a hardware or software VoIP phones, the major considerations remain the same:

• What VoIP call control protocols does the phone support?
  • H.323
  • SIP
  • MGCP
  • IAX2
• What VoIP codecs does the phone support?
  • G.711
  • G.722
  • G.723
  • G.726
  • G.727
  • G.728
  • G.729
  • ILBC
  • Speex
  • GSM - Full Rate
  • GSM - Enhanced Full Rate
  • GSM - Half Rate
  • DoS FS-1015
• Does the phone support 3-way calling
• Does the phone support Do-Not-Disturb (DND)
• Does the phone support custom ringtones?
• Does the phone provide a method to work behind routers and NAT?
• Does the phone support STUN?
• Does the phone support Symmetric RTP?
• Does the phone support a SIP outbound proxy?
• Does the phone support QoS
• Does the phone support encryption?
  • Secure RTP
  • AES

Choosing a Hardware VoIP Phone

When selecting a hardware VoIP phone, you should consider these items:

• What connections does the VoIP phone support?
  • Ethernet
    • Does the phone support Power Over Ethernet?
  • Wi-Fi
  • Dialup
  • ISDN
• Does the phone support IPv6?
• Does the phone support videoconferencing?
• Is the phone handset corded or cordless?
• Does the phone have a handset or a headset?
• Does the phone have a speakerphone?
• Does the phone have an LCD display?
  • Is the LCD display backlit?
• Does the phone have good ergonomics?
• Do you like the style of the phones?

Choosing a Software VoIP Phone

If you choose a software VoIP phone, you should consider these items:

• Does the phone software support my Operating System?
• Is the phone software easy to use?
• Does the software support customizable skins?
• Does the software support videoconferencing?
• Does the software support shared whiteboarding?

And, of course, the final purchasing decision should always include price as a criteria.

IAX

IAX is a call control protocol for VoIP.
IAX was designed to replace the earlier call control protocols, H.323 and SIP.
IAX is much more bandwidth efficient than the competing VoIP call control protocols, enabling it to support more concurrent VoIP calls over the same amount of bandwidth.

IAX traffic uses UDP port 4569. The use of a single well-known port enables IAX to be compatible with NAT (Network Address Translation), which can be a serious difficulty for earlier VoIP call control protocols.
IAX supports authentication using RSA public keys with the SHA-1 message digest algorithm for digital signatures.
IAX was developed for the Asterisk PBX and originally stood for Inter-Asterisk eXchange. IAX is now supported by many other VoIP platforms.

Common VoIP Hardware

VoIP hardware falls into several categories:

• VoIP Interface Cards for PCs
• PC Telephones
• VoIP Telephones
• VoIP Switches
• VoIP Gateways
• VoIP Routers
• VoIP PBX's
• VoIP Telephones

VoIP Interface Cards for PCs

VoIP Interface cards for PCs turn your PC into a very capable VoIP telephone.
Leading manufacturers of VoIP interface cards for the PC include:

• Digium
• VoiceTronix
• Quicknet

PC Telephones

PC Telephones are telephones which attach to your PC, usually via the USB port, and allow you to make telephone calls through your PC.

VoIP Telephones

VoIP telephones are telephones which attach directly to Ethernet network ports.

VoIP Switches

VoIP switches are devices which allow you to connect multiple phone lines to one Ethernet port. This allows every telephone which is connected to the switch to place VoIP calls.

VoIP Gateways

VoIP Gateways connect VoIP networks to the PSTN (Public Switched Telephone Network).

VoIP Routers

VoIP Routers route VoIP traffic in much the same way that regular routers route IP (Internet Protocol) traffic.

VoIP PBX's

VoIP PBX's are high-tech low-cost equivalents of traditional telephone PBX's. In addition to traditional PBX functionality, VoIP PBX's configure and manage VoIP network capabilities.

IP PBX

A PBX (Private Branch Exchange) is a small telephone switch owned by a company or organization. An IP PBX is simply a PBX which supports VoIP (Voice over IP). An IP PBX can also be referred to as a VoIP PBX.

An IP PBX may support VoIP both internally and externally. Internal VoIP support means that the IP PBX uses VoIP to communicate with each of its connect PBX phones. External VoIP supports means that the IP PBX uses VoIP to route calls to the outside world.
Most IP PBX's also support older analog or digital PBX phones and also support external connections on the public switched telephone network (PSTN).

How to Become a VoIP Reseller

If you are serious about reselling Voice over Internet Protocol (VoIP) services, there are some questions you will need to ask yourself first. Here are some simple guidelines to help you determine if and how you should pursue your goal of becoming a VoIP reseller.

Know the Service

If you really want to be a reseller for VoIP services, you need at least a basic working knowledge of how VoIP works and what type of applications are currently commonly used. Among the things you will need to understand are gateways and how they interact with voice switches. You will also need to understand the process for creating an integrated voice package that allows easy switching to and from conventional digital switches. Educate yourself on the basics before you attempt to move on to the next step--reselling.

Determine the Applications You Want to Sell

You may want to market VoIP to audio teleconferencing companies as a cost efficient means of participation during conference calls from any location. You may want to focus on providing Fortune 500 companies with a VoIP telephone service that virtually eliminates long distance charges. By selecting the types of applications you want to resell, you set the stage for moving on to your next step, which is becoming an agent or reseller.

Decide Whose Services You Want to Resell

Once you know what applications you want to resell, it is easy to begin investigating the companies that offer those types of services. Look into such qualities as reliability, customer support, private labelling options (if you want to sell under your own company name), and the rates offered. You may also want to see if billing and receiving payments are something you will have to do, or whether your supplier handle those functions for you. Don't be afraid to ask questions if you can't find documentation to specifically address a concern of yours. Companies that rely on resellers to generate revenue typically are very happy to work with persons who are serious and can think for themselves.
Being a VoIP reseller is an excellent way to make a living; and also a career choice that should be secure for a number of years to come. Investigate this possibility in more detail. You may find that this opportunity is right for you.

H.323

H.323 is an ITU standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks.

H.323 was the first standard for VoIP, but is being supplanted by SIP.
H.323 defines five components of a multimedia network:

• Terminals
• Multipoint Control Units (MCUs)
• Gateways
• Gatekeeper
• Border Elements

Terminals are telephone and PC equipment which connect end-users to the H.323 network.
MCUs are responsible for managing conferences. MCU's consist of a Multipoint Controller (MC) and an optional Multipoint Processor (MP). The MC manages signaling and the MP manages media mixing and switching.
Gateways nterface the H.323 network with other networks, including PSTN (Public Switched Telephone Network) and other H.323 networks. Gateways consist of a Media Gateway Controller (MGC) and a Media Gateway (MG). The MGC is is responsible for call signaling functions and the MG is responsible for media-related functions.
Gatekeepers are responsible for admission control and address resolution. Gatekeepers are able to provide advanced services such as normally found in PBX's.
Border Elements are positioned between two H.323 networks and assist in call routing and call authorization.

VoIP Conference Software

VoIP conference software comes in two basic varieties: free and commercial.
The free VoIP conference software packages tend to be difficult to setup and use; the commercial VoIP conference software packages tend to be slick and easy to install and use.
Of course, the commercial VoIP conference software packages also come with setup fees and recurring costs for conference room access.
Which option you choose depends upon your budget of time and money.

• Free VoIP Conference Software
• Commercial VoIP Conference Software

GnomeMeeting

GnomeMeeting is an H.323 compatible videoconferencing and VOIP/IP-Telephony application that allows you to make audio and video calls to remote users with H.323 hardware or software (such as Microsoft Netmeeting). It supports all modern videoconferencing features, such as registering to an ILS directory, gatekeeper support, making multi-user conference calls using an external MCU, using modern Quicknet telephony cards, and making PC-To-Phone calls.

Gspeakfreely

Gspeakfreely is a VoIP system with a flexible component system. It implements a set of audio processing components which can be connected to each other or mixed together. The most important components are net in/output, which implement VoIP functionality and the OSS-DSP in/output component.
Additionally there is a ISDN in/output component that allows making actual phone connections, and a file input component that can also play Internet radio streams. Also included is a fading plug-in, that can for example fade incoming calls into your music. New components can be developed for specific purposes, and combined with existing ones.
The net in/output components also have conference support. The net input component can mix incoming audio data from different hosts.

Commercial VoIP Conference Software

cu-HearMe

cu-HearMe allows any two or more computers in the world to be linked in a live, interactive voice environment via the web.
cu-HearMe supports both Microsoft Windows and MacOS.
cu-HearMe charges a setup fee and a monthly fee for conference room access.

Voice Now

Scheduling private conversations and conference calls is now as easy as visiting a web page. Voice Now's VoiceTech Communicator is a ready-to-run software application that has been designed to enhance existing public Internet and Voice over IP (VoIP) networks to include scheduled and instant conferencing services.
VoiceTech Communicator users can schedule their own IP based discussions or conference calls and then call in using a personal computer. VoiceTech's software-only architecture provides business conferencing with an unparalleled degree of flexibility at a reasonable cost.
Voice Now charges an annual fee for their service.

Icon Communicator

Icon Communicator is a modular software program which supports voice, conferencing, text-chat, and desktop sharing.
Icon Communicator is priced with a small setup fee and a monthly recurring fee for access to the Conference Centers.
The Icon Communicator is sold with an odd MLM (Multi-Level-Marketing) sales model. Theoretically, you can earn money by using and recommending this service to others.

Analog Telephone Adapter

An Analog Telephone Adapter (ATA), also known as the Analog Telephony Adapter, is an electronic device used to enable one or more analog telephones or facsimile machines for Voice over Internet Protocol (VoIP) calls and faxes. An Analog Telephone Adapter basically creates a physical connection by use of telephone and internet cables between a conventional phone or fax and a computer or an Ethernet gateway. The ATA usually comes with a digital phone and internet plan provider but it can also be bought independently. The Analog Telephone Adapter makes voice calls and faxing over the internet possible without the user needing to upgrade existing traditional telephony systems.

Benefits of Using an Analog Telephone Adapter

Making voice calls and sending faxes over the internet are significantly cheaper than doing so over traditional phone lines. There is no loss of functionality as call forwarding, call conferencing, and other features may still be included in VoIP subscription plans.
The Analog Telephone Adapter allows the user to immediately take advantage of the lesser costs involved in making calls over the internet. It also eliminates the need to get rid of existing phones and replace them with IP specialized phones. Moreover, ATAs are also generally cheaper than specialized digital VoIP phones which are capable of direct USB port or Ethernet gateway connectivity.

FXS to USB Adapters

The first type of Analog Telephone Adapters is called FXS to USB Adapters. FXS stands for Foreign Exchange Station. Such an ATA is a simple device that has one or more RJ-11 jacks or FXS telephone ports. Each phone jack can accommodate a phone or a fax machine. This ATA has an output cable which is in turn plugged into the personal computer's USB port. The computer becomes its means of connecting to the internet.
In this case, the ATA merely makes voice calls over the internet more convenient than using computer speakers and microphone or headsets; it does not directly interact with the VoIP server. The actual analog-to-digital conversion is done by the computer software, generally known as the softphone, which needs to be installed in the computer to which the ATA is connected. This software is supplied by VoIP service providers. This software converts voice data to digital data packets transmittable over the internet.

FXS to Ethernet Gateways

The second type of Analog Telephone Adapters directly performs analog-to-digital voice conversion. As such, Analog Telephone Adapters of this type do not need a softphone for VoIP. It communicates directly with the VoIP server using protocols such as SIP (the most common protocol for ATAs), H.323, IAX or MGCP. Voice signals are encoded and decoded using GSM, A-law, u-law and other such voice codecs.
Physically, an FXS to Ethernet Gateway ATA has one or more standard telephone jacks to which conventional telephony equipment can be plugged in. Analog voice data is converted to digital data and transported through an RJ-45 cable which is connected to the Local Area Network through its direct connection to the Ethernet hub or switch.

SIP

SIP (Session Initiation Protocol) is an IETF standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks.

SIP is an open standard replacement for the ITU's H.323.
SIP is described in RFC 3621 - SIP: Session Initiation Protocol.
SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.
SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.
Other RFC's which document SIP include:

• RFC 3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
• RFC 3263 - Session Initiation Protocol (SIP): Locating SIP Servers
• RFC 3264 - An Offer/Answer Model with the Session Description Protocol (SDP)
• RFC 3265 - Session Initiation Protocol (SIP)-Specific Event Notification

RTP

RTP (Real-Time Transport Protocol) is used to encapsulate VoIP data packets inside UDP packets.

RTP is defined in RFC 3550 - RTP: A Transport Protocol for Real-Time Applications.

RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers.
Other RFCs which document RTP include:

• RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control
  
• RFC 2032 - RTP Payload Format for H.261 Video Streams
• RFC 3158 - RTP Testing Strategies
• RFC 3605 - Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)
• RFC 3611 - RTP Control Protocol Extended Reports (RTCP XR)
• RFC 3711 - The Secure Real-time Transport Protocol (SRTP)

MGCP

MGCP (Media Gateway Control Protocol) is a protocol used within a Voice over IP (VoIP) system. This internal protocol was primarily developed to address the demands of carrier-based IP telephone networks. MGCP is a complementary protocol for both H.323 and SIP, which was designed as an internal protocol between the Media Gateway Controller and the Media Gateway. In MGCP, an MGC primarily handles all the call processing by linking with the IP network through constant communications with an IP signaling device, for example an SIP Server or an H.323 gatekeeper.
MGCP is comprised of a Call Agent, one MG (media gateway) which performs the conversion of media signals between circuits and packets, and one SG (signaling gateway) when connected to the PSTN (Public Switched Telephone Network). MGCP is widely used between elements of a decomposed multimedia gateway. The gateway has a Call Agent, which is comprised of the call control "intelligence" and a media gateway boasting the media functions, for example conversion from TDM voice to Voice over IP.
Media Gateways feature endpoints for the Call Agent to create and manage media sessions with other multimedia endpoints. Endpoints are sources and/or sinks of data that can be physical or virtual. For creating physical endpoints, hardware installation is needed while virtual endpoint can be created using available software.
Call Agents come with the capability of creating new connections, or modify an existing connection. Generally, a media gateway is a network element which provides conversion between the data packets carried over the Internet or other packet networks and the voice signals carried by telephone lines. The Call Agent provides instructions to the endpoints to check for any events and - if there is any - create signals. The endpoints are designed in such a way as to automatically communicate changes in service state to the Call Agent. The Call Agent can audit endpoints and the connections on endpoints.

MGCP Connections

MGCP connections can be point to point or multipoint. Point to point connection can be a connection between two endpoints for transmitting data between these endpoints. Once the connection is setup between two endpoints, data transfer takes place between the endpoints. In a multipoint connection, the connection is set up between an endpoint and a multipoint session. In a multipoint connection, connections can be created over various types of bearer networks.

MGCP Architecture

MGCP came to be a much sought after application of VoIP technology because it is not involved in the frustrating work of encoding, decoding, and transferring voice signals. Though, the MGCP Call Agent works as a software switch for a VoIP network, it really does nothing more than simply direct the media gateways and signaling gateways which perform all the work. One of the main tasks in building a Call Agent is implementing the numerous possibilities in the protocol. There are various informational RFCs which may explain the expected behavior under a wide range of conditions.

In MGCP architecture, each and every command features a transaction ID, gets an acknowledgement and receives a response. This can be better understood as subscription architecture, as the Call Agent informs the media gateways and the signaling gateways as to what events it takes care of and what events it leaves unattended.
MGCP packets are generally found wrapped in UDP port 2427. Similar to what you might find in TCP protocols, MGCP datagrams are formatted with whitespace. An MGCP packer can either be a command or a response. Commands start with a four-letter verb while "responses" start with a three number response code.

VoIP Codecs

A codec (Coder/Decoder) converts analog signals to a digital bitstream, and another identical codec at the far end of the communication converts the digital bitstream back into an analog signal.
In the VoIP world, codecs are used to encode voice for transmission across IP networks.
Codecs for VoIP use are also referred to as vocoders, for "voice encoders".
Codecs generally provide a compression capability to save network bandwidth. Some codecs also support silence suppression, where silence is not encoded or transmitted.

Codec	Algorithm	Bit Rate (Kbps)	Comments
ITU G.711	PCM (Pulse Code Modulation)	64	G.711 with mu-law used in North America and Japan, while G.711 with A-law used in the rest of the world.
ITU G.722	SBADPCM (Sub-Band Adaptive Differential Pulse Code Modulation)	48, 56 and 64
ITU G.723	Multi-rate Coder	5.3 and 6.4
ITU G.726	ADPCM (Adaptive Differential Pulse Code Modulation)	16, 24, 32, and 40
ITU G.727	Variable-Rate ADPCM	16-40
ITU G.728	LD-CELP (Low-Delay Code Excited Linear Prediction)	16
ITU G.729	CS-ACELP (Conjugate Structure Algebraic-Code Excited Linear Prediction)	8
ILBC	Internet Low Bitrate Codec	13.33 and 15.20
Speex	CELP (Code Excited Linear Prediction)	2.15-44.2	Part of the GNU Project and available under the Xiph.org variant of the BSD license
GSM - Full Rate	RPE-LTP (Regular Pulse Excitation Long-Term Prediction)	13
GSM - Enhanced Full Rate	ACELP (Algebraic Code Excited Linear Prediction)	12.2
GSM - Half Rate	CELP-VSELP (Code Excited Linear Prediction - Vector Sum Excited Linear Prediction)	11.4
DoD FS-1016	CELP (Code Excited Linear Prediction)	4.8

VoIP Gateway

A VoIP Gateway, or Voice over IP Gateway, is a network device which helps to convert voice and fax calls, in real time, between an IP network and Public Switched Telephone Network (PSTN). It is a high performance gateway designed for Voice over IP applications. Typically, a VoIP gateway comes with the ability to support at least two T1/E1 digital channels. Most VoIP gateways feature at least one Ethernet and telephone port. Controlling a gateway can be done with the help of the various protocols like MGCP, SIP or LTP.

Benefits of VoIP Gateways

The main advantage of VoIP gateway is that it can provide connection with your existing telephone and fax machines through the traditional telephone networks, PBXs, and key systems. This makes the process of making calls over the IP network familiar to VoIP customers.
VoIP gateways can end a call from the telephone and can provide user admission control using IVR (Interactive Voice Response) system and provide accounting records for the call. Gateways also help direct outbound calls to a specific destination, or can end the call from another gateway and send the call to the PSTN.
VoIP gateways plays a major role in enhancing carrier services and also supports the simplicity of the telephone calls for less cost and easy access. Flexible call integration has been developed at less cost which enables programmable call progress tones and distinctive ring tones.

Functions of VoIP Gateways

The main functions of VoIP gateways include voice and fax compression or decompression, control signaling, call routing, and packetization. VoIP gateways are also power packed with additional features such as interfaces to external controllers like Gatekeepers or Softswitches, network management systems, and billing systems.

Future of VoIP Gateway Technology

Over the years, VoIP gateway has become an efficient and flexible solution and is used for office data and voice connectivity. Besides the connectivity performance, VoIP also offers better reliability under a variety of circumstances.
The future of VoIP gateway is very clear and precise; high-density, scaleable, open platforms need to be designed and implemented to allow the millions of installed telephones and fast-growing number of H.323 computer clients (such as Netscape's Communicator and Microsoft's NetMeeting) to communicate over IP. Many vendors are in the process of designing interoperable VoIP gateways according to the latest architectures to meet the changing demands of service providers, corporate network clients, and individual carriers.

What is VoIP?

VoIP (Voice over Internet Protocol) is simply the transmission of voice traffic over IP-based networks.
The Internet Protocol (IP) was originally designed for data networking. The success of IP in becoming a world standard for data networking has led to its adaption to voice networking.

The Economics of VoIP

VoIP has become popular largely because of the cost advantages to consumers over traditional telepone networks. Most Americans pay a flat monthly fee for local telephone calls and a per-minute charge for long-distance calls.

VoIP calls can be placed across the Internet. Most Internet connections are charged using a flat monthly fee structure.
Using the Internet connection for both data traffic and voice calls can allow consumers to get rid of one monthly payment. In addition, VoIP plans do not charge a per-minute fee for long distance.
For International calling, the monetary savings to the consumer from switching to VoIP technology can be enormous.

VoIP Telephones

There are three methods of connecting to a VoIP network:

• Using a VoIP telephone
• Using a "normal" telephone with a VoIP adapter
• Using a computer with speakers and a microphone

Types of VoIP Calls

VoIP telephone calls can be placed either to other VoIP devices, or to normal telephones on the PSTN (Public Switched Telephone Network).
Calls from a VoIP device to a PSTN device are commonly called "PC-to-Phone" calls, even though the VoIP device may not be a PC.
Calls from a VoIP device to another VoIP device are commonly called "PC-to-PC" calls, even though neither device may be a PC.

How does VoIP work?

VoIP or Voice Over Internet Protocol (sometimes called Internet Telephony) is touted in some circles as the technology of future. The reasoning is simple, really. VoIP is bringing possibilities to the forefront of technological thinking because the possibilities were listed as impossible just a few years ago. VoIP uses a broadband Internet connection for routing telephone calls, as opposed to conventional switching and fiberoptic alternatives. This process holds great promise in providing higher efficiency and lower cost for communication consumers. One interesting aspect of the technology is that, for the user, no large scale infrastructure is required. It's all about combining the functionality of the Internet and a conventional phone into one single service with minimal software and hardware support.

 

How Does it Work?

The most common way VoIP works is that the end user establishes a hi speed broadband connection, a router and a VoIP gateway. Instead of a standard telephone line, the router sends the telephone calls over an Internet connection. The VoIP gateway, placed somewhere in direct proximity of the connected Internet converts the analog signals into digital format, which are further broken down into smaller chunks called 'packets', before sending it over the Internet, much like the way data is transmitted to and from the computer. These packets are sent to their final destination and instructions for bringing back into an understandable form are embedded in them. It then goes through a VoIP gateway where the packets are reconverted into the original analog format utilizing a PSTN (Public Switched Telephone Network), thereby routing the call to the number the caller has dialed blending old school technology and hi tech delivery in a seamless and instantaneous way.

More Than One Way to Make a Call

Using VoIP technology, phone calls can also be made using IP phones between two computers. IP phones looks like normal standard handsets, but equipped with an RJ-45 Ethernet connector in place of the common RJ-11 connectors. These phones come with all the necessary hardware and software pre-loaded, allowing the user to directly connect to the router bringing the new user into the cost effective world of VoIP.
PC to PC calls are the easiest and most inexpensive way to make use of VoIP technology. There are many companies providing software for free or at reduced cost to encourage consumer experimentation with VoIP. When calling from a PC, all the user may need is a microphone, a suitable sound card and a reliable Internet connection. The service itself may be free of cost in many cases. The only fee the end user may have is the monthly fee for the Internet service provider and nothing additional for the actual calls made.

VoIP Features

The biggest advantage of VoIP is that the customers can make calls from anywhere in the world where a broadband Internet connection is available. The customers can take their IP phones or ATA's with them on national and international trips and still can manage to access what is essentially an individual's domestic phone line.
Then there are the softphones, which a software application that loads the VoIP services onto the desktop or laptop. Some even simulate an interface that looks like a telephone, with which you can place VoIP calls to anybody around the world, through a standard broadband connection.
Most VoIP services come with the caller id, call waiting, call transfer, repeat dialing and three-way dialing features. For additional features such as call filtering, forwarding a call, or sending calls directly to the voice mail, the service provider may assess an additional fee. Most VoIP services also allowthe user to check his/her voicemail over the web or attach messages to an e-mail that is sent to his/her PDA or PC.
Generally, the facilities and components provided by VOIP phone system suppliers and service operators may vary in significant ways. It is advisable to check the pros and cons before subscribing. Make sure that you have available technical support for the possible compatibility issues that could arise between the existing and new hardware components.

Conclusion

VoIP is still in its infancy. While it holds great promise, it has some major technical hurdles to jump, such as emergency calling, and the need for an uninterruptible power source (i.e. PC battery backup). However, as VoIP is set to become more widely available, let's hope there will be reliable solutions in place for the existing problems in the coming years. Who knows? In another five years, we may have VoIP system sans a router and the VoIP service being more common than conventional phone networks we rely on so heavily today.

How to Setup Skype

Voice over Internet Protocol, or VoIP for short, is quickly becoming recognized as one of the leading ways of communication over the Internet.
Skype is one of the most widely used programs for PC to PC calling using VoIP. To use this, you will need to have Skype installed on your computer.
You can download skype from Get Skype.
A window will open asking what you want to do with a file called SkypeSetup.exe. Click 'Run' to downloads and execute the software. A confirmation message may appear asking if you're sure you want to run this software. Just click 'Run' again.
Skype enables you to talk to anyone around the world for free. However there are a few requirements that must be met by both ends calling. This consists of a few key bits of equipment:

• A broadband Internet connection
• A microphone
• Speakers
• Skype

When the download has finished, the Skype Setup Wizard will appear and guide you through the rest of the installation. The installation is self explanitory. Accept the terms of service, and click "next". Skype should begin to install - it should take no less than 10 seconds.
Depending on what version you have downloaded, it will give you the choice to install a toolbar for Internet explorer. This is not essential, and is your own choice. After the installation wizard has finished, there should hopefully be a new icon on your desktop - a white S inside a light blue circle - This is the Skype Icon.
Upon running Skype for the first time, you will be confronted with a login dialogue box. This is where you will enter your Skype login details. If you do not have an account, just click "Don't have a Skype name" underneath the Username field.
This will cause another dialogue box to appear, with several more fields; asking you for your name, your desired Skype name, and your desired password. All this information is kept confidential. After filling these out, and accepting the terms and conditions, click next. Another dialogue box pops up asking for your current email address and your Country and city. After filling out the desired fields, hit "Sign in" and you should be signed in.
After you have successfully signed in for the first time, a "getting started" guide will appear. This guide will help you to configure your microphone and speakers so that optimum performance can be achieved. Following the on-screen instructions, you should not encounter any problems.
To test your set-up, go to the main Skype window. In the contacts section, there should be 1 contact, the Skype test call. By calling this number (double clicking) you can listen to what your microphone sounds like, and what to expect from a call.
Once you are satisfied with your sound, the set-up is complete. You can now easily access Skype and use it to make calls anytime, without having to pay a cent.

Free VoIP

Free VoIP calls are never completely free of cost. Often, the goal is not to achieve completely free calls to all destinations, but to use the VoIP operator that suits your needs best. Keeping that in mind, you will learn that most VoIP companies will let you talk for free in their own network, but will charge you for making calls outside their proprietary network.

Companies offering free VoIP calls usually offer free calls inside their own network and towards other specially selected destinations. Using this tactic users are drawn to make calls to free destinations and later purchase credits to make calls towards paid destinations.
There are several options to choose from, when it comes to selecting the right VoIP service for you.

1. The first category of VoIP services involves direct PC to PC connections, or PC to landline/mobile phone connections.
This is by far the most common way of using VoIP. It requires the use of VoIP software. Skype is one of the most popular VoIP services on the market in this category. You can initiate conversations with other Skype users free of charge. However, if you want to make calls to regular landlines or mobile phones, you will need to pay. The subscription fee for calls in North America is $30 a year. It is not a great deal of money, but it still isn't free.
Raketu, another player in the VoIP market, offers free phone calls to landlines in 42 countries and besides that, it also offers live video television. The only downside to Raketu's service is that they ask you to pay $9.95 up front in order to use their services.
It you are still looking for a cost-free VoIP service, you can use something like voipCheap that allows you to make free calls to PCs and regular phone lines. However, this service is available up to the maximum limit of 300 minutes per week, per IP address, after which you will have to pay. It includes many destinations outside USA and Canada that can be called without paying a cent, but if you want to call places not on their free calls list, you will be charged.
Google has recently come up with a new VoIP service called Google Voice, which is now being tried by many users in the US. At present, users can sign up by invitation only. Free calls and messages can be sent anywhere in the mainland US, and international calls can be made for less than half a dollar per minute.

2. The second category of VoIP services involves getting landline or mobile phone connections to be routed over the Internet.
Jajah is a good example from this category. Users of this service can make free calls to landlines and mobiles in select countries. For other select countries in Europe, Asia and South America, you can apply for landline calls.
The YouFon service routes cellular phone calls from Nokia E-Series handsets through the Internet. It lets you make free calls to other registered YouFon users for costs just around 5 dollars a month.

3. A third category of VoIP technology services involves the use of VoIP phones directly using an Internet Protocol (IP) network by connecting to it via a Wi-Fi or Ethernet network.
The ooma service is a good example from this category for users within the US. It uses peer-to-peer VoIP technology for users to make free calls within the US. The service is available for residents and up to a limit of 3000 minutes of outgoing calls. The only cost that is involved is in purchasing the device. Certain premium services are available for an extra charge.
If you manage to use services that offer good deals for what you require, you can save up to 98% from your phone bills. This means that you will need to put some time and research into it, but in the end you can achieve nearly free calls via the VoIP technology.