VoIP (voice over IP – that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service VoIP is therefore telephony using a packet based network instead of the PSTN (circuit switched).
During the early 90’s the Internet was beginning its commercial spread. The Internet Protocol (IP), part of the TCP/IP suite (developed by the U.S. Department of Defense to link dissimilar computers across many kinds of data networks) seemed to have the necessary qualities to become the successor of the PSTN.
The first VoIP application was introduced in 1995 – an “Internet Phone”. An Israeli company by the name of “VocalTec” was the one developing this application. The application was designed to run on a basic PC. The idea was to compress the voice signal and translate it into IP packets for transmission over the Internet. This “first generation” VoIP application suffered from delays (due to congestion), disconnection, low quality (both due to lost and out of order packets) and incompatibility.
VocalTec’s Internet phone was a significant breakthrough, although the application’s many problems prevented it from becoming a popular product. Since this step IP telephony has developed rapidly. The most significant development is gateways that act as an interface between IP and PSTN networks.
What is VoIP?
Voice over IP (VoIP) is a blanket description for any service that delivers standard voice telephone services over Internet Protocol (IP). Computers to transfer data and files between computers normally use Internet protocol.
Voice over IP is the technology of digitizing sound, compressing it, breaking it up into data packets, and sending it over an IP (internet protocol) network where it is reassembled, decompressed, and converted back into an analog wave form..
The transmission of sound over a packet switched network in this manner is an order of magnitude more efficient than the transmission of sound over a circuit switched network.
As mentioned before, VoIP saves bandwidth also by sending only the conversation data and not sending the silence periods. This is a considerable saving because generally only one person talks at a time while the other is listening. By removing the VoIP packets containing silence from the overall VoIP traffic we can reach up to 50% saving. In a circuit switched network, one call consumes the entire circuit. That circuit can only carry one call at a time.
In a packet switched network, digital data is chopped up into packets, sent across the network, and reassembled at the destination. This type of circuit can accommodate many transmissions at the same time because each packet only takes up what bandwidth that is necessary.. Internet Telephony simply takes advantage of the efficiencies of packet switched networks.
Gateways are the key component required to facilitate IP Telephony. A gateway is used to bridge the traditional circuit switched PSTN with the packet switched Internet. The gateway allows the calls to transfer from one network to the other by converting the incoming signal into the type of signal required by the network it is required to send it on. For example, A PC user wishes to call someone using a conventional phone. The PC sends the IP packets containing digitized voice to the gateway.
The software package chosen will reflect the organizational needs, but should contain the following modules as defined in the Technology Guide Series – Voice Over IP Publication, and other sources.
- Voice Processing Module. This aspect of the software is required to prepare voice samples for transmission. The functionality provided by the voice processing module should support:
- A PCM Interface is required to receive samples from the telephony interface (e.g. a voice card) and forward them to the Voice Over IP software for further processing.
- Echo Cancellation is required to reduce or eliminate the echo introduced as a result of the round trip exceeding 50 milliseconds.
- Idle Noise Detection is required to suppress packet transmission on the network when there are no voice signals to be sent. This helps to reduce network traffic as up to 60% of voice calls are silence and there is no point in sending silence.
- A Tone Detector is required to discriminate between voice and fax signals by detecting DTMF (Dial Tone Multi frequency) signals.
- The Packet Voice Protocol is required to encapsulate compressed voice and fax data for transmission over the network.
- A Voice Playback Module is required at the destination to buffer the incoming packets before they are sent to the Codec for decompression.
- Call Signaling Module. This is required to serve as a signaling gateway which allows calls to be established over a packet switched network as opposed to a circuit switched network (PSTN for example).
- Packet Processing Module. This module is required to process the voice and signaling packets ready for transmission on the IP based network.
- Network Management Protocol. Allows for fault, accounting and configuration management to be performed.
The exact hardware, which would be required, again, depends on organizational needs and budget. The list below highlights the most general hardware required.
- The most obvious requirement is the existence (or installation) of an IP based network within the branch office gateway is required to bridge the differences between the protocols used on an IP based network and the protocols used on the PSTN.
- The gateway takes a standard telephone signal and digitizes it before compressing it using a Codec. The compressed data is put into IP packets and these packets are routed over the network to the intended destination.
- The PC’s attached to the IP based network require the voice/fax software outlined above. They also require Full Duplex Voice Cards which allow both communicating parties to speak at the same time – as often happens in reality.
- As an alternative to installing Voice Cards, IP Telephones can be attached to the network to facilitate Voice Over IP. A secondary gateway should be considered as a backup in the event of the failure of the primary gateway.
There are many protocols in existence but the main ones are considered to be the following:
- H.323 is an ITU (International Telecommunications Union) approved standard which defines how audio /visual conferencing data is transmitted across a network. H.323 relies on the RTP (Real-Time Transport Protocol) and RTCP (Real Time Control Protocol) on top of UDP (User Datagram Protocol) to deliver audio streams across packet based networks.
- G.723.1 defines how an audio signal with a bandwidth of 3.4KHz should be encoded for transmission at data rates of 5.3Kbps and 6.4Kbps. G.723.1 requires a very low transmission rate and delivers near carrier class quality. The VoIP Forum as the baseline Codec for low bit rate IP Telephony has chosen this encoding technique.
- G.711.The ITU standardised PCM (Pulse Code Modulation) as G.711. This allows carrier class quality audio signals to be encoded for transmission at data rates of 56Kbps or 64Kbps. G.711 uses A-Law or Mu-Law for amplitude compression and is the baseline requirement for most ITU multimedia communications standards.
- Real-Time Transport Protocol(RTP) is the standard protocol for streaming applications developed within the IETF (Internet Engineering Task Force).
- Resource Reservation Protocol(RSVP) is the protocol which supports the reservation of resources across an IP network. RSVP can be used to indicate the nature of the packet streams that a node is prepared to receive.