|Voice over IP (Voip, pronounced voyp) is a technology that allows voice conversations to be conducted over the Internet (hence IP, for Internet Protocol) instead of the plain old telephone system(POTS). |
The basic principle of Voip is very simple. It's the same technology you have probably used already to listen to music over the Internet. Voice sounds are picked up by a microphone and digitized by the sound card. The sounds are then converted to a compressed form, compact enough to be sent in real time over the Internet, using a software driver called a codec. The term codec is short for "encoder/decoder". The sounds are encoded at the sending end, sent over the Internet and then decoded at the receiving end, where they are played back over the speakers. The only requirements are a connection between the two computers of an adequate speed, and matching codecs at each end.
To be usable, a Voip system also needs a method for establishing and managing a connection, for example, calling the other computer, finding out if they accept the call, and closing the connection when a user hangs up. Because Voip allows two way communication, and even conference calls, it's a lot more complicated than simple audio streaming. How calls are managed is the area in which Voip systems fundamentally differ, and two Voip users must be using the same system (or compatible ones) in order to be able to call each other.
Because most Internet users don't have a permanent Internet address (IP address, a number like 184.108.40.206 that uniquely identifies that computer, at that moment), Voip systems don't generally work by calling another computer direct – although that may be an option for those who do have a permanent address. Instead, each user of the service registers with an intermediate server, which maintains a record of their IP address all the time they are connected. An example of a Voip application that works this way is Picophone. The small size of the PicoPhone application file (about 64Kb, barely larger than Windows Notepad) demonstrates clearly that the basic principles of Voip are not
complicated to implement.
Another reason for using an intermediate server is that it eases the problem of getting Voip to work through the firewalls that everyone uses these days. Many firewalls block any data from the Internet that is not sent in response to a specific request. This makes it impossible to call another computer direct. Because the called computer did not request any data from the caller, the call request would be blocked. By establishing a connection with a server, the Voip software opens a channel of communication through which other computers can call it. Communication may continue using the server, or information may be passed via the server that allows the two computers to open a direct connection between them and continue using that.
| Although the basic requirements of a Voip system are quite simple, real-world implementations are quite complex. Voip systems in widespread use today fall into three groups: systems using the H.323 protocol, systems using the SIP protocol, and systems that use proprietary protocols. On our services we provide SIP standard SIP (for Session Initiation Protocol) is an Internet Engineering Task Force (IETF) standard signalling protocol for teleconferencing, telephony, presence and event notification and instant messaging. |
It provides a mechanism for setting up and managing connections, but not for transporting the audio or video data. It is probably now the most widely used protocol for managing Internet telephony.
Like all IETF protocols, SIP is defined in a number of RFCs (Request For Comments – the standards documents that define Internet standard protocols) principally RFC 3261. A SIP-based Voip implementation may send the encoded voice data over the network in a number of ways. Most implementations use Real-time Transport Protocol (RTP), which is defined in RFC 3550. Both SIP and RTP are implemented on UDP which, as a connectionless protocol, can cause difficulties with certain types of routers and firewalls. Usable SIP phones therefore also need to use STUN (for Simple Traversal of UDP over NAT), a protocol defined in RFC 3489 that allows a client behind a NAT router to find out its external IP address and the type of NAT device.
Cisco Catalyst switches offer leading voice-aware services that integrate with the Cisco Unified Communications family of products to make it easier to deploy, operate, and consolidate voice solutions. As a result, our users get access to new, high-quality voice applications .
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Secure IP Communications everywhere
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Faster resolution of voice issues