Tuesday, 6 November 2012

Voice Over IP

VoIP (voice over IP) is an IP telephony term for a set of facilities used to manage the delivery of voice information over the Internet. VoIP involves sending voice information in digital form in discrete packets rather than by using the traditional circuit-committed protocols of the public switched telephone network . A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service.

Voice over IP, commonly refers to the communication protocols and transmission techniques involved in the delivery of voice communications and multimedia sessions over Internet Protocol. Internet telephony refers to communication services: voice, fax, sms and voice messaging applications, that are transported over INTERNET rather than Public Switched Telephone Network(PSTN).

The steps involved in originating a VOIP telephone call are signaling and media channel setup, digitization of analog voice signal, encoding, packetization and transmission as INTERNET protocol over a packet switched network.

On receiving side, similar steps reproduce the original voice stream. VOIP is one of the technology used by IP telephony to transport phone calls.

Voice Over IP has been implemented in various ways using different protocols:
Media Gateway control Protocol
Session Initiation Protocol
Real Time Transport Protocol
Session Description Protocol

In addition to IP, VoIP uses the real-time protocol to help ensure that packets get delivered in a timely way. Using public networks, it is currently difficult to guarantee Quality of Service. Better service is possible with private networks managed by an enterprise or by an Internet telephony service provider.

Using VoIP, an enterprise positions a "VoIP device" at a gateway. The gateway receives packetized voice transmissions from users within the company and then routes them to other parts of its intranet (local area or wide area network) or, using a T-carrier system or E-carrier interface, sends them over the public switched telephone network.


The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.
RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number.
RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol which assists in setting up connections across the network.


The Session Initiation Protocol (SIP) is a signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multi-party (multicast) sessions. Sessions may consist of one or several media streams.
Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games.
The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol(SCTP). It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).


The Session Description Protocol (SDP) is a format for describing streaming media initialization parameters. SDP is intended for describing multimedia communication sessions for the purposes of session announcement, session invitation, and parameter negotiation. SDP does not deliver media itself but is used for negotiation between end points of media type, format, and all associated properties. The set of properties and parameters are often called a session profile. SDP is designed to be extensible to support new media types and formats.

SDP started off as a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with Real-time Transport Protocol (RTP), Real-time Streaming Protocol (RTSP), Session Initiation Protocol (SIP) and even as a standalone format for describing multicast sessions.

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