Category Archives: VoIP

Obtaining Free VoIP

VoIP stands for Voice over Internet Protocol. This delivers voice transmissions over IP networks. One important thing to understand about “free” VoIP calls is they are not totally free. Even if you would subscribe to a VoIP provider that claims it would not charge you anything, keep in mind that you still have to pay for your broadband Internet connection.

Your goal in using VoIP is not to obtain free calls to all destinations, but to have an operator that would best suit your needs. Keeping that in mind, you will know that most VoIP companies will enable you to have free talks in their own network, but will charge you for the calls that you made for the proprietary network.

Free calls within the network of companies offering free VoIP calls, as well as to other specially chose destinations, often serve as a come on to users. This entices them use the free calls, earn credits to call paid destinations, and pay the call charges not covered by the credits.

There are a number of methods employed by these companies to attract clients and make a profit at the same time. The big thing about VoIP calls is they are very inexpensive, although not totally free.

Skype is one of the most popular VoIP service providers today. With it, you can start conversations with other computer users of Skype without any cost. With Skype software, you can have free phone calls from a computer to another computer. Should you, however, call regular landlines, you will be charged. In North America, the subscription fee for Skype is $30 every year. This is not really big money for the whole year, but it’s still not free. If you have to contact someone who has no computer or Internet connection, you will have to pay the required cost.

Raketu is another provider of VoIP service. It offers its consumers free phone calls to landlines in more than 42 countries; it also offers live video presentations. The catch is that you have to pay $9.95 to utilize their free services. Some say that this amount of money is used as a subscriber’s credit just in case he or she happens to call unlisted numbers. Nevertheless, you still pay for a service which is claimed to be “free”.

For more information on obtaining free VoIP read:

  • Free VoIP
  • Free VoIP
  • Free VoIP
  • Free VoIP
  • Skype

    Skype is a software that is now commonly used over the Internet to transfer calls, whether with or without a fee. Aside from making telephone calls either to landlines or mobile phones, users of Skype service can also avail of other features such as instant messaging, video conferencing, or file transfer.

    Skype was started by a team of entrepreneurs and software developers. Their headquarters is based in Luxembourg, but they have other branch offices in major cities like London, Prague, Stockholm, Tallinn, Tartu, and San Jose.

    Since the initial launching of this service, Skype has seen rapid growth and is also becoming a popular choice for instant messaging along with Yahoo! Messenger and MSN.

    Skype Features

    1. SkypeIn
    2. This feature enables Skype users to receive calls on their computer by simply dialing a regular phone through a local Skype phone number. There are local Skype numbers available for various countries in Asia, Australia, Europe, South America, and the United States.

      You can use this local number for each country and you will be charged a similar rate with standard calls within the country.

    3. Videoconferencing
    4. This feature was launched in January of 2006 for both Mac and Windows. The Skype 2.0 for Linux also features support for Videoconferencing. Meantime, Skype for Windows is capable of “high-quality” video with full-screen or screen-in-screen modes.

    5. Skype on mobile device
    6. In April of this year, Skype has announced that it will be available in over 50 mobile phone networks worldwide. Aside from that, Skype has also been made available on the PSP (PlayStation Portable) on Slim Lite model although you will need one of three microphone input peripherals.

      Other mobile devices in which Skype are also available include the following:

      • Windows Mobile
      • N800 and N81 Internet Tablets
      • Blackberry

    Skype is also equipped with a Secure Communication feature that is not visible to the user. Its registration system requires no proof of identity; hence users can avail of the service without revealing their identity. If you have tried signing up for a Skype account, you realize how easy it is to set up an account using any name without any proof of authenticity.

    However, there are also ways of controlling how you use the Skype services. They are as follows:

    • Do not launch the program unless you are ready to use it. When expecting a call, verify it through other means.
    • Refrain from keeping your calls lengthy.
    • As soon as you finished with the call, quit the application. However, you must turn it off instead of merely closing the application window.

    MGCP

    Media Gateway Control Protocol (MGCP) is a procedure used in VoIP (Voice over IP) systems. This was created to cater to the needs of carrier-based IP telephony networks. MGCP is a protocol that corresponds to H.323 and SIP, devised as an internal system between the Media Gateway (MG) and Media Gateway Controller (MGC).

    An MGC manages all call processing by connecting to the IP network through continuous communication with an IP signaling tool. An example of such is an H.323 gatekeeper or SIP server.

    MGCP is composed of one Call Agent (CA), MG (to perform media signal conversion between packets and circuits), and Signaling Gateway (SG). They all connect to a PSTN (Public Switched Telephone Network). MGCP is mostly within a decomposed multimedia gateway. This gateway has a CA composed of the call control ‘nucleus’ and a media gateway which operates media functions.

    MGs have multimedia endpoints with which the CA creates and manages media sessions with other endpoints. Endpoints are data sources or data sinks that can be either virtual or physical. Hardware installation is required to generate physical endpoints while creating virtual endpoints need software installation.

    CAs have the capacity to produce new connections or alter an existing one. Broadly, a media gateway is a component that offers conversion between Internet data packets (or other network packets) and voice transmissions carried by phone lines. The CA gives instructions to endpoints to detect events and create signals. Endpoints intend to convey variations in service state to the CA mechanically. The CA then examines endpoints and the associations between endpoints.

    VoIP Gateway

    A VoIP Gateway, or Voice over IP Gateway, is a network device that converts fax and voice calls between an Internet Protocol (IP) network and a public switched telephone network (PSTN). This device bridges conventional telephone equipment and networks to telephone networks that use the VoIP technology.

    The types of VoIP Gateways are analog units and digital units.

    Functions of VoIP Gateways

    Its principal functions include:

    • Call routing,
    • Fax/voice compression/decompression
    • Control signaling
    • Packetization

    Other functions include interfaces to external controllers, such as network management systems, billing systems, and Gatekeepers or Soft Switches.

    Majority of VoIP gateways have at least one telephone port and one Ethernet port. Protocols like SIP, LTP, or MGCP help in controlling a gateway.

    Advantages of VoIP Gateways

    One of the major advantages of a VoIP gateway is that it can provide connection to your present fax machines and telephone through the traditional key systems, PBXs, and telephone networks. This makes the procedure of making calls through the IP network familiar to VoIP users.

    VoIP gateways can finish a telephone call and can give user admission control with the Interactive Voice Response (IVR) system. They can also present the call’s accounting records. They will help lead outgoing calls to a specific destination, or finish the call from another gateway and give a call to the PSTN.

    VoIP gateways have a crucial role in developing carrier services. They also support the ease of the telephone calls for trouble-free access and less cost. Compliant call integration is enhanced at a lesser amount to allow distinctive ring tones and programmable call progress tones.

    Potentials of VoIP in the Future

    For many years, VoIP gateway has proven to be a proficient and adaptable solution in voice connectivity and office data. Aside from its good connectivity operation, VoIP also provides better dependability under various situations.

    The potentials of VoIP in the near future are very apparent and clear-cut. Scaleable, open, and high-density platforms have to be developed and employed to enable the millions of telephones and the rapidly escalating number of H.323 computer clients (such as MS NetMeeting and Netscape Communicator) to keep in touch over IP.

    A number of VoIP developers are currently designing interoperable VoIP gateways based on the most recent architectures. They aim to cater to the shifting demands of individual carriers, corporate network clients, and service providers.

    RSVP

    RSVP refers to Resource Reservation Protocol, a protocol used in VoIP to manage Quality of Service or QoS. QoS pertains to a set of algorithms that aim to provide different quality levels to different types of network traffic. RSVP is a part of the IIS or Internet Integrated Service model, which provides controlled-link sharing and real-time services.

    Essentially, RSVP functions by making a request to reserve a specific bandwidth and latency. These resources are requested by the RSVP through the VoIP telephone call. RSVP makes this request to every network device between the two VoIP units. Using RSVP, an individual can reserve bandwidth prior to a specific program via the Internet. He is then able to receive the program at a higher rate.

    RSVP is a protocol that supports unicast (one source to one receiver) and multicast (one source to many receivers) signaling. It is capable of setting up and maintaining information about reservation states. These pieces of information are located at each router through which a specific stream of data is being transferred.

    RSVP is used by a number of systems for specific purposes. Host computers use RSVP to request Qualities of Service from the network. The Quality of Service will then be used by the host for specific application data streams. Routers also use RSVP to request Quality of Service requests to all nodes where the data flow will pass through. This in turn allows routers to establish and maintain the proper network state. Through RSVP, each node on the path of the data flow will reserve an amount of resource from the system.

    VoIP

    Voice over Internet Protocol (VoIP) is a transmission protocol facilitating voice transmission over IP-based networks and other packet-switched networks. The standard Internet Protocol (IP) has given way for adaptation to voice networking.

    VoIP facilitates a number of tasks and offers services that may be harder and more costly to execute using the Public Switched Telephone Network (PSTN). It can transmit more than one telephone call over the same broadband connection. This allows an easier addition of extra telephone lines to an office or home. It also facilitates call forwarding, caller ID, and conference calling at a minimal cost (none or nearly none, unlike traditional telecommunication companies).

    Cost Reduction

    VoIP technology is very popular because it offers compelling features that make switching from the other system more tempting. The number one advantage of using VoIP technology over traditional telephone networks is cost reduction.

    The calls in VoIP can be placed across the Internet. Majority of the Internet connections are paid in a fixed monthly fee arrangement; so if the internet connection is utilized for both voice and data traffic calls, clients can eliminate one monthly payment. Moreover, plan VoIP structures do not charge the consumer for each minute of a long-distance call.

    Getting Rid of Phone Lines

    Using VoIP can let you call off your traditional phone service through your local telephone company. You can have all your telephone calls done over your Internet connection.

    A residential consumer can save more or less $40 each month, while consumers involved in business can save thousands of dollars each month.

    Getting Rid of Long Distance Charges

    With VoIP, a consumer can also save on costs with long-distance calls. Most of the business and residential telephone clients pay per-minute charges for long-distance telephone calls. VoIP technology can minimize or get rid of these long-distance fees. This savings are particularly helpful to clients frequently making international calls charged per minute.

    Number Portability

    Using VoIP technology, a consumer can easily take his or her number with him. For example, if a person who has a phone number in Denver moves to Miami, he or she can still keep the Denver number. This is very helpful for families and acquaintances to contact one another wherever they go.

    VoIP Codec

    A codec (coder-decoder or compressor-decompressor) is a program used to convert voice signals into the digital bit stream and back to be conveyed over the Internet or any network during a VoIP call. The codecs for VoIP are also termed “vocoders,” meaning voice encoders.

    Codecs typically perform encoding-decoding, compression-decompression, and least often, encryption-decryption.

    Encoding – Decoding

    When you use a conventional public switched telephone network (PSTN) phone, the produced voice is carried in an analog manner over the phone line. When VoIP technology is used, the voice is converted into digital signals. This conversion is the process of encoding. When this digital signal reaches its end point, it has to be decoded back to its original analog state so the other party can perceive sound and understand it.

    Compression – Decompression

    Bandwidth is a limited article of trade so if the data to be transmitted is less bulky, you can transmit more in a given time, hence enhancing its function. To make the digitized data (the voice) lighter, it is compressed by the codec.

    Compression is an intricate course of action whereby the same data is saved but there is less digital space used. During this process, the digitized data is restrained to a packet to the compression algorithm.

    The lighter data is transmitted over the IP network. Once it arrives at its end point, it is decompressed back to its previous form before it becomes decoded. Most of the time, the data is not decompressed back since the compressed data is already in a consumable state.

    Encryption – Decryption

    Encryption is one of the best ways to provide security. In encryption, the data is converted into a state which no one can understand. This way, even if the encrypted data is interrupted by illicit individuals, the data would still be confidential. When the encrypted data gets to its end point, it is decrypted back to its previous state. Sometimes, data is already somehow encrypted when it is compressed, since it is changed from its original form.

    There are many codecs for video, audio, text, and fax. As a user, you may deem it unnecessary for you to study what these are. However, it is always good to know even a little of these codecs, since you might have to relate codecs concerning VoIP in your business in the future.

    For more information on VoIP Codec read:

  • VoIP Codec
  • VoIP Codec
  • Free VoIP Software Phones

    VoIP, or Voice over Internet Protocol, is a cost-effective answer to your telephone demands. It enables you to make low-priced calls to anyone, wherever you may be in the world, with the use of your VoIP phone handset or your PC. Moreover, calls within the network of the provider are free of charge.

    Here is a list of free VoIP Software phones available in the market.

    Wengophone

    Wengophone is an SIP (Session Initiation Protocol) phone enabling its users to communicate from a computer to other users of SIP accommodating VoIP software at no cost. It also enables its users to call cell phones and landlines, make video calls, and send SMS messages. None of these functions require a specific SIP provider. This means they can be used with any provider present in the market, unlike proprietary solutions like Skype.

    Speak Freely

    This phone is a 100% free Internet telephone first designed by John Walker in 1991. After April 1996, he stopped developing the program. Since then, various other Internet phones have been produced across the globe. However, most of these programs charge its users. Majority of these programs have poor sound quality and do not support the fundamental features of Speak Freely such as selectable compression, the answering machine, and encryption.

    Ekiga

    Previously known as GnomeMeeting, this VoIP phone is an H.323 compatible VoIP-IP-Telephony and video conferencing application. It enables the user to make video and audio calls to other users with H.323 hardware or software, such as MS NetMeeting and Netscape Communicator. Ekiga supports all video conferencing features to make multi-user conference calls with an external MCU. It also allows the user to make computer-to-phone calls, use modem Quicknet telephony cards, and register to an ILS directory. It also provides gatekeeper support.

    Twinkle

    Twinkle is a phone for VoIP communications using the Session Initiation Protocol. You can use Twinkle in a network using an SIP proxy to route your calls or for direct IP phone to IP phone conversations.

    Aside from making indispensable voice calls, this phone also offers an astounding number of features, namely:

    • 3-way conference calling;
    • call waiting;
    • multiple active call identities;
    • 2 call lines;
    • blind call transfer;
    • reject call redirection request;
    • out-of-band DTMF (SIP INFO);
    • user definable number conversion rules;
    • STUN support for NAT traversal;
    • RFC 2833 DTMF events;
    • automatic failover to an alternate server if a server is unavailable;
    • call redirection when busy, no answer, on demand, unconditional; and
    • a whole lot more

    Other free VoIP software phones include but are not limited to:

    • linphone;
    • minisip;
    • OhPhone;
    • Internet Switchboard;
    • SIPSet;
    • Kphone; and
    • Jabbin.

    Additional Resources on VoIP Software Phones