Entry

Glossary Description

AMR

Adaptive Multi-Rate (AMR) is an Audio data compression scheme optimized for speech coding. AMR is adopted as the standard speech codec by 3GPP. The codec has eight bit rates, 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbit/s. The bitstream is based on frames which contain 160 samples of 20 milliseconds long. More...

CNG

Comfort noise generation (CNG) fills in the silent portions of VAD, namely processed transmissions with artificial noise ("comfort noise"). This means a low volume level appropriate for the average volume level of the received signals. CNG is compliant with the G711 codec standard. More...

EVRC

Enhanced Variable Rate CODEC - EVRC is a speech codec used by CDMA networks. It was developed in 1995 to replace QCELP. The codec supports three source rates of 9.6 kbit/s (full rate), 4.8 kbit/s (half rate) and 1.2 kbit/s (eight rate). More...

G711

G711 is an ITU-T speech codec for audio companding, a method of reducing the effects of a channel with limited dynamic range. Companding is a combination of compressing and expanding and is a variant of audio level compression. G711 encoders create 64 kbit/s bit streams. More...

G722.2 (AMR-WB)

Adaptive Multi Rate - WideBand or AMR-WB. AMR-WB operates like AMR with various bit rates, namely 6.60, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 and 23.85 kbits/s. The codec provides excellent speech quality due to wider speech bandwidth of 50 - 7000 Hz. More...

G723.1a

G723.1 is an audio codec for voice that compresses voice audio in chunks of 30 milliseconds. A look-ahead of 7.5 ms duration is also used. G723.1A is the implementation of the ITU G723.1 standard. It uses ACELP and MP-MLQ for coding at 5.3 kbps and 6.3 kbps respectively. More...

G726

G726 is ITU-T speech codec operating at bit rates of 16-40 kbit/s. The most commonly used mode is 32 kbit/s, half the rate of G711, and thus can increase the usable network capacity by 100%. The standard is based on ADPCM technology. More...

G728

G728 is a ITU-T standard for speech coding operating at 16 kbit/s. G728 passes low bit rate modem signals of up to 2400 bit/s, in addition to network signalling. The complexity of the codec is 30 MIPS, and 2 kBytes of RAM is needed for codebooks. More...

G729AB and G729E

G729 is an audio data compression algorithm for voice that compresses voice audio into chunks of 10 milliseconds. G729a is compatible with G729, but requires less computation. G729b is a silence compression scheme, which has a VAD module used to detect voice activity, speech or non speech. G729E is the ITU-T recommendation G729, but modified to achieve a bit rate of 11.8 kbps. More...

GSM-AMR-NB

GSM-AMR-NB is an Adaptive Multi Rate-Narrow Band (AMR-NB) speech codec standard with eight basic bit rates, 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbit/s. This codec works on the ACELP principle for all bit rates, and supports DTX, VAD and CNG algorithms.

GSM-EFR

Enhanced Full Rate or EFR or GSM-EFR is a speech coding standard that was developed to improve the quality of the GSM-Full Rate (FR) codec. At 12.2 kbit/s, the EFR provides wirelike quality under both noise free and background noise conditions. EFR is compatible with the highest AMR mode. More...

GSM-FR

GSM-FR is based on RPE-LTP (Regular Pulse Excitation - Long Term Prediction) speech coding paradigm. Linear prediction is used in the synthesis filter with an order of 8. By contrast, narrowband speech codecs have an order of 10, and wideband speech codecs, an order of 16. More...

iLBC

The Internet Low Bit Rate Codec (iLBC) is a royalty free narrowband speech codec, developed by Global IP Sound (GIPS). It is suitable for VoIP applications, streaming audio, archival and messaging. The algorithm is a version of block-independent linear predictive coding, with the choice of data frame lengths of 20 and 30 milliseconds. The encoded blocks have to be encapsulated in a suitable protocol for transport, eg. RTP.iLBC is defined in RFC 3951. It is one of the codecs used by both Gizmo Project and Skype. More...

QCELP13

QCELP is a speech codec developed in 1994 to increase the speech quality of the IS-96A codec used earlier in CDMA networks. QCELP is also known as Qualcomm PureVoice. QCELP13 is one version of QCELP, using 13kbits/s. The other version is QCELP8, using 8kbits/s. More...

SMV

Selectable Mode Vocoder - SMV is a speech codec standard providing multiple modes of operation, and is used in CDMA-2000 networks. SMV for Wideband CDMA is based on 4 codecs: full rate at 8.5 kbit/s, half rate at 4 kbit/s, quarter rate at 2 kbit/s, and eighth rate at 800 bit/s. More...

VAD

Voice activity detection or voice activity detector - VAD is an algorithm used in speech processing, wherein the presence or absence of human speech is detected from audio samples. When used for speech coding, VAD is compliant with the G711 codec standard. More...

WMA 9

Windows Media Audio (WMA) is a proprietary compressed audio file format developed by Microsoft. It was initially a competitor to the MP3 format, but has positioned itself as a competitor to the Advanced Audio Coding format used by Apple and is part of the Windows Media framework. WMA 9 is a bundled version that includes three more codecs, namely a voice codec, lossless codec and the WMA 9 Pro codec. The Pro is based on a completely different technology. The most current version of the format is Windows Media Audio 9.1. More... More...

Entry

Glossary Description

H261

H261 is an 1990 ITU video coding standard originally designed for transmission over ISDN lines on which data rates are multiples of 64 kbit/s. The data rate of the coding algorithm was designed to be able to operate between 40 kbit/s and 2 Mbit/s. H261 was the first practical digital video coding standard. The H261 design was a pioneering effort, and all subsequent international video coding standards (MPEG-1, MPEG-2/H262, H263, and even H264) have been based closely on its design. More...

H263

H263 is a video codec designed as a low-bitrate encoding solution for videoconferencing. It was initially intended to be utilized in H324 based systems, but has since found use in RTP/IP-based videoconferencing, H320-based videoconferencing, RTSP (streaming media) and SIP (Internet conferencing). More...

H264/MPEG-4 AVC

H264, MPEG-4 Part 10, or AVC, for Advanced Video Coding, is a digital video codec standard which is noted for achieving very high data compression. The intent of H264/AVC project has been to create a standard that would be capable of providing good video quality at bit rates that are substantially lower (e.g., half or less) than what previous standards would need. More...

MPEG 4 (Simple Profile)

MPEG-4 is the designation for a group of audio and videocodingstandards and related technology agreed upon by the ISO/IEC Moving Picture Experts Group (MPEG). The primary uses for the MPEG-4 standard are for streaming media and CD distribution, videophone and broadcast television. More...

WMV 9

Windows Media Video (WMV) is a generic name for the set of video codec technologies developed by Microsoft. It is part of the Windows Media framework. The codecs were originally developed as proprietary codecs for low-bitrate streaming applications. However, in 2003 Microsoft drafted a video codec specification based on its Windows Media Video version 9 codec and submitted it to SMPTE for standardization. The standard was officially approved in March 2006 as SMPTE 421M. More...

Entry

Glossary Description

CID

Caller ID(Caller Identity Display or CID) is a telephony intelligent network service that transmits the caller's telephone number to the called party's telephone. Typically, CID is transmitted digitally using Bell 202 modulation between the first and second rings. Also known as Calling Line Identification (CLI) when provided via an ISDN connection to a PABX. More...

DTMF

Dual-tone multi-frequency (DTMF), also known as Touch Tone or Tone Dialing, is used for over the line telephone signaling to the call switching center via the voice frequency band. DTMF is an example of a multi-frequency shift keying (MFSK) system. More...

MF-R1 & MR-R2

Multi-Frequency (MF) is an outdated, in-band signaling technique. MF is the precursor of DTMF tones, the well known "touch tones". In contrast to R2_Signalling, MF signaling is sometimes referred to as R1. R2 is mnemonic for Region Two signaling (Europe) to differentiate it from R1 signaling, the North American MF signaling. More...

RFC 2833

RFC 2833is the IETF standard that describes RTP payload for DTMF digits, telephony tones, and telephony signals. RTP packets carry dual-tone multi-frequency (DTMF) signaling, other tone signals and telephony events.

WMV 9

Windows Media Video (WMV) is a generic name for the set of video codec technologies developed by Microsoft. It is part of the Windows Media framework. The codecs were originally developed as proprietary codecs for low-bitrate streaming applications. However, in 2003 Microsoft drafted a video codec specification based on its Windows Media Video version 9 codec and submitted it to SMPTE for standardization. The standard was officially approved in March 2006 as SMPTE 421M. More...

Entry

Glossary Description

G168

G168 is ITU-T's Digital Network Echo Canceller. Applications include removing acoustic echoes from a full-duplex conferencing system. Typically, echo tails ranging between 16ms to 128ms can be digitally subtracted from a circuit echo.

Entry

Glossary Description

RTP

Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was originally designed as a multicast protocol, but has since been applied in many unicast applications. It goes along with the RTP Control Protocol (RTCP) and is built on top of the User Datagram Protocol (UDP). More...

RTCP

Real Time Control Protocol (RTCP) provides out-of-band control information for an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP. More...

UDP

User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. It is a minimal message-oriented transport layer protocol that is currently documented in IETF RFC 768. Using UDP programs on networked computers can send short messages known as datagrams to one another. More...

Entry

Glossary Description

T30

T30 is an ITU standard handshake protocol for G3 facsimile communication over IP. The standard also manages the fax session, error correction and information exchange between two fax devices.

T38

T38 is the ITU-T recommendation for fax over IP (based) networks in real time. The recommendation defines a real time method for faxing over IP networks as it supports the use of either TCP or UDP.

V17

V17 is an ITU-T fax protocol that uses TCM modulation at 12 and 14.4 kbit/s. CCITT analog facsimile, analog modem, signaling standard, providing up to 14400 kbps data rates and backwards compatible to the V29 standard, which supports speeds of up to 9600 kbps.

V21

V21 is an ITU-T recommendation for full-duplex communication between two analogue dial-up modems using audio frequency-shift keying modulation at 300 bauds to carry digital data at 300 bit/s. It is a variant of the originalBell 103 modulation format.More...

V27, V27bis V27ter

V27 is an ITU-T recommendation for full-duplex or half-duplex communication between two analogue fixed-line modems. It uses PSK modulation at 1600 bauds to carry synchronous data at 4800 bit/s. The V27bis extension of V27 added a fall-back modulation rate. The V27ter extension was defined for use on dial-up lines.More...

V29

V29 is an ITU-T standard for fax operations that specifies speeds up to 9,600 bps with fallback to 7,200 bps.

V34, V34HD

V34 is an ITU-T recommendation for a modem, allowing up to 28.8 kbit/s bidirectional data transfer. Other additional defined data transfer rates are 24.0 kbit/s and 19.2 kbit/s as well as all the permitted V32 and V32bis rates.More...

Entry

Glossary Description

V92

V92 is an ITU-T recommendation that establishes a modem standard supporting near 56 kbit/s download and 48 kbit/s upload rates. It is intended to succeed the V90 standards. With V92, PCM is used for both the upstream and downstream connections; previously 56K modems only used PCM for downstream data. More...

V150.1, V8 Gateway

The V150 standards provide selectable options for some of the layers of the Modem over IP Gateway. First, the standard specifies two types of compatible Gateways - Universal Modem Relay Gateways and V8 Gateways. A Universal Modem Gateway (U-MR) will perform full termination of a specific set of V-series modulations. A V8 Gateway provides termination for modulations negotiated through V8. White Paper...

Entry

Glossary Description

Bell103 /
Bell 212

The Bell 103 modem was the first commercial modem for computers, released by AT&T in 1962. It allowed digital data to be transmitted over regular telephone lines at a speed of 300 bits per second. The Bell 103 modem used audio frequency-shift keying to encode data. Different pairs of audio frequencies were used by each station. Bell 212 succeeded the Bell 103 in delivering 1200 bps in each direction. See also V22.More...

Frequency-
shift
Keying (FSK)

Frequency-shift keying (FSK) is a form of frequency modulation in which the modulating signal shifts the output frequency between predetermined values. Usually, the instantaneous frequency is shifted between two discrete values termed the mark frequency and the space frequency. This is a noncoherent form of FSK. More...

MNP2-4,
MNP5

Microcom Networking Protocol (MNP) is a communications protocol that provides data compression and error connection in modems. More...

HDLC for PPP/
RFC 1662

High-Level Data Link Control (HDLC) is a bit-oriented synchronous data link layer protocol developed by the International Organization for Standardization (ISO). HDLC is now the basis for synchronous PPP (point to point protocol) used by many servers to connect to a wide area network, most commonly the Internet.The RFC 1662 specification deals with PPP in HDLC-like Framing. More...

V21

V21is an ITU-T recommendation for full-duplex communication between two analog dial-up modems using audio frequency-shift keying modulation at 300 bauds to carry digital data at 300 bit/s. It is a variant of the original Bell 103 modulation format. More...

V22
& V22bis

V22 is an ITU-T recommendation for full-duplex communication between two analog dial-up modems. It uses PSK modulation at 600 baud to carry data at 1200 or 600 bit/s. V22bis is an ITU-T recommendation that extends V22 with a faster rate. It uses QAM modulation at 600 baud to carry digital data at 2400 or 1200 bit/s. The 1200 bit/s mode is compatible with V22. More...

V23

V23 is an ITU-T recommendation for half-duplex communication between two analog dial-up modems using FSK modulation at up to 600 or 1200 bauds to carry digital data at up to 600 or 1200 bit/s respectively. An optional 75 bauds reverse channel carries 75 bit/s. More...

V32 &
V32bis

V32is an ITU-T recommendation for a modem, enabling bidirectional data transfer at either 9.6 kbit/s or 4.8 kbit/s at a symbol rate of 2,400 baud instead of the 600 baud of the V22 standards. V32bis is an ITU-T recommendation for a modem, allowing up to 14.4 kbit/s bidirectional data transfer. More...

V34
(Synchronous)

V34 is an ITU-T recommendation for a modem, allowing up to 28.8 kbit/s bidirectional data transfer. Other additional defined data transfer rates are 24.0 kbit/s and 19.2 kbit/s as well as all the permitted V32 and V32bis rates. Most V34 modems support V.FC, although all modern modems support both. More...

V42 &
V42bis

V42 is an error correction protocol promoted by the ITU-T. Its function is to enable a receiver to immediately request re-transmission of lost data packets. V42 turns an error-prone communications path into an error-free path, and is generally included in dialup modems as well. V42bis reduces the amount of sent data modulated, being that modulation and bandwidth are the major bottlenecks in today's dial-up connections. More...

V44

V44 is an adaptive data compression standard incorporated into the V92 dial-up modem standard. V44 offers somewhat better compression performance for certain types of data than the V42bis standard, on average allowing 5% greater throughput. V44 is based on LZJH (Lempel-Ziv-Jeff-Heath) compression technology. More...

V90

V90 is an ITU-T recommendation for a modem, allowing 56 kbit/s download and 33.6 kbit/s upload. It was developed between March 1998 and February 1999. It is also known as V.Last as it was anticipated to be the last standard for modems operating near the channel capacity of POTS lines to be developed. A follow-on standard, V92, was developed later in 1999 to replace V90. More...

V92

V92is an ITU-T recommendation, titled Enhancements to Recommendation V90, that establishes a modem standard allowing near 56 kbit/s download and 48 kbit/s upload rates. V92 was first presented in August 1999. It is intended to succeed the V90 standards. With V92 PCM is used for both the upstream and downstream connections; previously 56K modems only used PCM for downstream data. More...

Entry

Glossary Description

IWF

The IWF (Inter-Working Function) connects the circuit-switched data paths of a Mobile Network with a Fixed Network (PSTN/ISDN). Powerpoint

A-TRAU
frame

A-TRAU frame: Rate adaptation scheme used to deliver data in a 14.4Kbps traffic channel between the Mobile Station/User Equipment and the IWF (Inter-Working Function).

HSCSD

High-Speed Circuit-Switched Data (HSCSD), is a development of Circuit Switched Data, the original data transmission mechanism of the GSM mobile phone system. As with CSD, channel allocation is done in circuit switched mode. The difference comes from the ability to use different coding methods and even multiple time slots to increase data throughput. More...

GSM
03.45

The GSM TS 03.45 standard is intended for transparent faxes and mobile-to-mobile calls.

RLP
Ver. 0,1,2

The Radio Link Protocol (RLP) is a layer 2 LAPB based protocol which performs grouping of user data for the purpose of implementing error control and retransmission mechanisms in the case of non-transparent low layer capabilities. The RLP layer is in charge of the transmission of the data compression parameters to the peer RLP entity and to the L2R layer, when those parameters have to be negotiated. The function that realizes the implementation of the protocol (described in 3GPP TS 24.022) takes place at both ends of the GSM connection in the MT and the IWF/MSC.

V110
(ITU)

An ITU standard for Rate Adaptation on ISDN. This allows a terminal adapter (TA) to connect to a low speed device (50bps to 19.2Kbps), such as a PC COM port, and convert the data so it can be sent over a 64Kbps link.

Entry

Glossary Description

3G-324M

3G-324M is a standard umbrella protocol for supporting multimedia transmission using 3G technologies. It consists of a signaling channel defined by H245 protocol. More...

H324

H324 is an ITU-T recommendation for voice, video and data transmission over regular analog phone lines. It uses a regular 33,600 bit/s modems for transmission, the H263 codec for video encoding and G723 for audio. More...

Entry

Glossary Description

CALEA

The U.S. Congress passed the Communications Assistance for Law Enforcement Act (CALEA) to aid law enforcement in its effort to conduct surveillance of citizens via digital telephone networks. The Act obliges telephone companies to make it possible for law enforcement agencies to tap any phone conversations carried out over its networks, as well as making call records available. More...

UPD/IP

User Datagram Protocol (UPD): A protocol within the TCP/IP protocol suite that is used in place of TCP when a reliable delivery is not required. There is less processing of UDP packets than there is for TCP. UDP is widely used for streaming audio and video, voice over IP (VoIP) and videoconferencing, because there is no time to retransmit erroneous or dropped packets.

VLAN

A virtual LAN, commonly known as a vLAN or as a VLAN, is a logically-independent network. A VLAN consists of a network of computers that behave as if connected to the same wire - even though they may actually physically connect to different segments of a LAN. Network administrators configure VLANs through software rather than hardware, which makes them extremely flexible. More...

Entry

Glossary Description

SIP

Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements. It is one of the leading signalling protocols for Voice over IP, along with H323. More...

H323

H323 is an umbrella recommendation from the ITU-T, that defines the protocols to provide audio-visual communication sessions on any packet network. It is a part of the H32x series of protocols which also address communications over ISDN, PSTN or SS7. H323 is commonly used in Voice over IP and IP-based videoconferencing. The alternatives to H323 are IETF's SIP, MGCP and IAX. More...

H248/
MEGACO

Media Gateway Controller (Megaco) is a signalling protocol used between a Media Gateway and a Media Gateway Controller (also known as a Call Agent or a Soft Switch) in a VoIP network. It defines the necessary signalling mechanism to allow a Media Gateway Controller (Call agent) to control gateways in order to support voice/fax calls between PSTN-IP or IP-IP networks. More...

MGCP

Media Gateway Control Protocol (MGCP) is a protocol used within a Voice over IP system and supersedes the Simple Gateway Control Protocol (SGCP). It is an internal protocol used within a distributed system that can appear to the outside world as a single VoIP gateway. More...