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What is Voice Over IP?

Voice over Internet Protocol, also called Voice over IP, VoIP, SMS (for short) is a group of resources that enable the voice signal to travel over the Internet using an IP (Internet Protocol). This means that sends the voice signal into digital form in packets rather than sending it (in digital or analog) through circuits useable for telephony only as a conventional PSTN telephone company (an acronym for Public Switched Telephone Network, Telephone Network Public Switched).

The protocols that are used to carry voice signals over the IP network are commonly referred to as Voice over IP protocols and IP protocols can be viewed as commercial implementations of the experimental Network Voice Protocol (1973), invented by ARPANET.

The Voice over IP traffic can travel on any IP network, including those connected to the Internet, such as local area networks (LAN).

It is very important to differentiate between voice over IP (VoIP) and Telephony over IP.

     * VoIP is the set of standards, devices, protocols, ultimately the technology that allows transmission of voice over IP protocol.
     * IP Telephony is the set of new functionalities of the phone is, as it turns the traditional phone services because they finally can get to be able to offer due to carry voice over IP data networks in.

[Note: Not to be confused with IP Telephony ToIP (Text-over-IP)]

Advantages

The main advantage of such services is that it avoids the high telephone charges (mostly long distance) that are customary for companies in the PSTN (PSTN). Some cost savings are due to using a single network to carry voice and data, especially when users have not used the full capacity of an existing network in which they can use VoIP without additional cost. VoIP calls from any VoIP provider are generally free, in contrast to the VoIP to PSTN calls generally costs the VoIP user.

The development of VoIP codecs (alaw, G.729, G.723, etc.) enabled encoded in the voice data packets from ever smaller. This leads to voice communications over IP require very narrow bandwidths. Along with the progress of ADSL connections in the residential market, this type of communication is very popular for international calls.

There are two types of PSTN to VoIP services: Direct Inbound Dialing (Direct Inward Dialing: DID) and access numbers. The DID connects the call directly to the VoIP user while access numbers require that user enters the extension number of the VoIP user. Access numbers are usually charged as a local call for those who made the call from the PSTN and are free to the VoIP user.

These prices can be up to 50 times cheaper than the prices of local operators.

Functionality

VoIP can facilitate tasks that would be more difficult to perform using the common telephone networks:

     * The telephone calls can be automatically routed to a VoIP phone, regardless of where it is connected to the network. One could carry a VoIP phone on a trip, and any site connected to the Internet, you could receive calls.
     * Toll-free number for use with VoIP is available in the United States of America, United Kingdom and other countries from organizations such as VoIP User.
     * Call center agents using VoIP phones can work anywhere with an Internet connection is fast enough.
     * Some packages include VoIP services for which extra PSTN (PSTN) typically charge a surcharge or that is not available in some countries, such as 3 calls at once, call return, redial automatic, or Caller ID.

Mobility

The users of VoIP can travel anywhere in the world and still make and receive calls as follows:

     * The services of the subscribers of telephone lines can make and receive calls outside their local area. For example, if a user has a phone number in the city of New York and traveling through Europe and someone calls your phone number, this will be received in Europe. Also, if a call is made from Europe to New York, will be charged as local call, of course if you travel to Europe must have an Internet connection available.
     * Users of Instant Messenger based VoIP services can also travel anywhere in the world and make and receive phone calls.
     * VoIP phones can integrate with other services available on the Internet, including video, data and exchange messages with other services in parallel with the conversation, audio conference, address book management and information exchange with others (friends, colleagues, etc. ).

Impact on Trade

Voice over IP is cheaper for international communications and thus improve communication between suppliers and customers, or between delegations from the same group.

Also the voice over IP is being integrated through specific applications in web portals. In this way, users can request a call from that company or a call to schedule a time in particular, to be launched through a Voice IP normally.

Future of Voice over IP

Bandwidth growing worldwide and optimize the computers of layer 2 and 3, to ensure the QoS (Quality of Services) of voice services in real time makes the future of Voice over IP is very promising. In United States providers of voice over IP like Vonage achieved a market share. In Spain, thanks to the flat rates for voice, conventional operators save the landing mass of these operators. However, the expansion of this technology is coming from the hand of the developers of systems such as Avaya and Cisco in their platforms that integrate voice and data networks. Other manufacturers such as Alcatel, Nortel Networks, Matra, Samsung and LG also develop corporate solutions for voice over IP in its telecommunications equipment.

Standard VoIP (H323)

Defined in 1996 by the ITU (International Telecommunication Union) to the various manufacturers provide a set of standards to enable them to evolve together.

VoIP is not a service, its a technology

In many countries of the world, VoIP has generated numerous disagreements between the territorial and legal advice about what this technology is clear and should be clear that technology is not a VoIP service as such, but a technology that uses the Internet Protocol (IP) through which compressed and decompressed in a highly efficient data packets or datagrams, to allow communication by two or more clients via a network like the Internet. With this technology you can provide telephony and videoconferencing, among others.

Network Architecture

The actual standard defines three key elements in its structure:

     * Terminals: These are replacements for existing phones that can be implemented in both software and hardware.

     * Gatekeepers: These are the core of any organization’s VoIP and would substitute for existing plants. Usually implemented in software, if any, all communications go through it.

     * Gateways: This is the link to the traditional telephone network that acts transparently to the user.

With these three elements, the structure of the VoIP network may be the connection of two delegations from the same company. The advantage is immediate: all communications between the delegations are completely free. This same scheme could be implemented for suppliers.

     * Protocols for VoIP: It's the language we use for different devices to connect VoIP. This part is important because it will depend on the efficiency and complexity of communication.
           By order of seniority (from oldest to newest):
                 + H.323 - Protocol defined by the ITU-T
                 + SIP - Protocol defined by IETF
                 Megace + (also known as H.248) and MGCP - Control Protocols
                 + Skinny Client Control Protocol - Protocol owned Cisco
                 + Minet - Protocol owned Mitel
                 Cornet-IP + - Protocol owned Siemens
                 + IAX - original protocol for communication between Asterisk PBXs (It is a standard for other data communications systems, currently in version 2 - IAX2)
                 + Skype - Protocol proprietary peer-to-peer application Skype is used in the
                 + IAX2 - Protocol for communication between replacing PBXs Asterisk IAX
                 + Jingle - open protocol used by Jabber technology
                 + MGCP Cisco-proprietary protocol
                 WeSIP + - Free Protocol License VozTelecom

VoIP, as we have seen a lot of advantages for both businesses and users. The question is why has not yet implemented this technology? Then we analyze the apparent reasons for that VoIP has not yet been imposed in conventional telephony.

Parameters for VoIP

The main problem today presents both the penetration of VoIP and IP for all applications. Ensuring quality of service over an IP network, through delays where bandwidth it is not possible, therefore, presents different problems in ensuring quality of service.

Codecs

The voice must be encoded to be transmitted by the IP network. This makes use of codecs that will ensure the encryption and compression of audio or video for subsequent decoding and decompressing before it can generate a sound or an image usable. Depending on the codec used in the transmission, it will use more or less bandwidth. The amount of bandwidth is usually directly proportional to the quality of data transmitted.

The VoIP codecs used are the G.711, G.723.1 and G.729 (specified by the ITU-T)

These codecs utilize the following payload bandwidth.

    * G.711 bit-rate of 56 or 64 Kbps
    * G.722 bit-rate of 48, 56 or 64 Kbps
    * G.723 bit-rate of 5.3 or 6.4 Kbps
    * G.728 bit-rate of 16 Kbps
    * G.729 bit-rate of 8 or 13 Kbps

 

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