H.324

Last updated
H.324
Terminal for low bit-rate multimedia communication
StatusIn force
Latest version(04/09)
Organization ITU-T
Related standards G.711, G.722, G.723.1, 3G-324M
Website https://www.itu.int/rec/T-REC-H.324

H.324 is an ITU-T recommendation for voice, video and data transmission over regular analog phone lines. It uses a regular 33,600 bit/s modem for transmission, the H.263 codec for video encoding and G.723.1 for audio.

ITU-T Technical Committee of the ITU

The ITU Telecommunication Standardization Sector (ITU-T) is one of the three Sectors of the International Telecommunication Union (ITU); it coordinates standards for telecommunications and Information Communication Technology.

A voice message is a message containing audio of a person's voice. Voice itself could be 'packaged' and sent through the IP backbone so that it reaches its marked 'address'. In a technical sense, the process of sending 'voice packets' is a semi passive way of communication. However, given the speed at which it could be delivered can make the communication sound seamless.

Video electronic medium for the recording, copying and broadcasting of moving visual images

Video is an electronic medium for the recording, copying, playback, broadcasting, and display of moving visual media. Video was first developed for mechanical television systems, which were quickly replaced by cathode ray tube (CRT) systems which were later replaced by flat panel displays of several types.

Contents

H.324 standard is formally known as Terminal for low bit-rate multimedia communication. H.324 covers the technical requirements for very low bit-rate multimedia telephone terminals operating over the General Switched Telephone Network (GSTN). H.324 terminals provide real-time video, audio, or data, or any combination, between two multimedia telephone terminals over a GSTN voice band network connection.

H.324 terminals offering audio communication shall support the G.723.1 audio codec. H.324 terminals offering video communication shall support the H.263 and H.261 video codecs. G.722.1 may be used for wideband audio applications. Annex G of H.324 specification defines usage of ISO/IEC 14496-1 (MPEG-4 Systems) generic capabilities in H.324 terminals. H.324/I terminals shall support interoperation with voice telephones using G.711 speech coding, if the connected network supports transmission and reception of G.711. Other modes such as G.722 audio may optionally be supported as well. [1]


H.261 is an ITU-T video compression standard, first ratified in November 1988. It is the first member of the H.26x family of video coding standards in the domain of the ITU-T Video Coding Experts Group, and was the first video coding standard that was useful in practical terms.

G.722.1 is a licensed royalty-free ITU-T standard audio codec providing high quality, moderate bit rate wideband (50 Hz – 7 kHz audio bandwidth, 16 ksps audio coding. It is a partial implementation of Siren 7 audio coding format developed by PictureTel Corp.. Its official name is Low-complexity coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss.

MPEG-4 is a method of defining compression of audio and visual (AV) digital data. It was introduced in late 1998 and designated a standard for a group of audio and video coding formats and related technology agreed upon by the ISO/IEC Moving Picture Experts Group (MPEG) under the formal standard ISO/IEC 14496 – Coding of audio-visual objects. Uses of MPEG-4 include compression of AV data for web and CD distribution, voice and broadcast television applications.

H.324 was adapted by 3GPP to form 3G-324M.

It is for example used in the Vialta Beamer BM-80 Phone Video Station, the MINX system from Datapoint Corporation, and in several other videophones.

See also

Related Research Articles

H.263 is a video compression standard originally designed as a low-bit-rate compressed format for videoconferencing. It was developed by the ITU-T Video Coding Experts Group (VCEG) in a project ending in 1995/1996 as one member of the H.26x family of video coding standards in the domain of the ITU-T, and it was later extended to add various additional enhanced features in 1998 and 2000. Smaller additions were also made in 1997 and 2001, and a unified edition was produced in 2005.

Speech coding is an application of data compression of digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.

G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames. An algorithmic look-ahead of 7.5 ms duration means that total algorithmic delay is 37.5 ms. Its official name is Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s. It is sometimes associated with a Truespeech trademark in coprocessors produced by DSP Group.

The Adaptive Multi-Rateaudio codec is an audio compression format optimized for speech coding. AMR speech codec consists of a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality speech starting at 7.4 kbit/s.

Adaptive Multi-Rate Wideband (AMR-WB) is a patented wideband speech audio coding standard developed based on Adaptive Multi-Rate encoding, using similar methodology as algebraic code excited linear prediction (ACELP). AMR-WB provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders which in general are optimized for POTS wireline quality of 300–3400 Hz. AMR-WB was developed by Nokia and VoiceAge and it was first specified by 3GPP.

G.729 is a royalty-free narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds. It is officially described as Coding of speech at 8 kbit/s using code-excited linear prediction speech coding (CS-ACELP). The wide-band extension of G.729 is called G.729.1, which equals G.729 Annex J.

G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new 16 kbit/s rate. The four bit rates associated with G.726 are often referred to by the bit size of a sample, which are 2, 3, 4, and 5-bits respectively. The corresponding wide-band codec based on the same technology is G.722.

Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB. It adds support for stereo signals and higher sampling rates. Another main improvement is the use of transform coding additionally to ACELP. This greatly improves the generic audio coding. Automatic switching between transform coding and ACELP provides both good speech and audio quality with moderate bit rates.


H.320 or Narrow-band visual telephone systems and terminal equipment is an umbrella Recommendation by the ITU-T for running Multimedia (Audio/Video/Data) over ISDN based networks. The main protocols in this suite are H.221, H.230, H.242, audio codecs such as G.711, and video codecs such as H.261 and H.263.

3G-324M

3G-324M is the 3GPP umbrella protocol for video telephony in 3G mobile networks.

The Video Coding Experts Group or Visual Coding Experts Group is a working group of the ITU Telecommunication Standardization Sector (ITU-T) concerned with video coding standards. It is responsible for standardization of the "H.26x" line of video coding standards, the "T.8xx" line of image coding standards, and related technologies.

H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.

Siren is a family of patented, transform-based, wideband audio coding formats and their audio codec implementations developed and licensed by PictureTel Corporation. There are three Siren codecs: Siren 7, Siren 14 and Siren 22.

G.719 is an ITU-T standard audio coding format providing high quality, moderate bit rate wideband audio coding at low computational load. It was produced through a collaboration between Polycom and Ericsson.

Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 80 Hz to 14 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz or even up to 22 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16-bits to encode samples, also resulting in much better voice quality.

G.718 is an ITU-T recommendation embedded scalable speech and audio codec providing high quality narrowband speech over the lower bit rates and high quality wideband speech over the complete range of bit rates. In addition, G.718 is designed to be highly robust to frame erasures, thereby enhancing the speech quality when used in internet protocol (IP) transport applications on fixed, wireless and mobile networks. Despite its embedded nature, the codec also performs well with both narrowband and wideband generic audio signals. The codec has an embedded scalable structure, enabling maximum flexibility in the transport of voice packets through IP networks of today and in future media-aware networks. In addition, the embedded structure of G.718 will easily allow the codec to be extended to provide a superwideband and stereo capability through additional layers which are currently under development in ITU-T Study Group 16. The bitstream may be truncated at the decoder side or by any component of the communication system to instantaneously adjust the bit rate to the desired value without the need for out-of-band signalling. The encoder produces an embedded bitstream structured in five layers corresponding to the five available bit rates: 8, 12, 16, 24 & 32 kbit/s.

Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required data bandwidth for a given signal-to-noise ratio.

References

  1. ↑ ITU-T (2005-09) (PDF) H.324 : Terminal for low bit-rate multimedia communication, Retrieved on 2009-07-21
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