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Encyclopedia > G.722

G.722[1] is a ITU-T standard wideband speech codec operating at 48-64 kbit/s. Technology of the codec is based on split band ADPCM. The ITU Telecommunication Standardization Sector (ITU-T) coordinates standards for telecommunications on behalf of the International Telecommunication Union (ITU) and is based in Geneva, Switzerland. ... A codec is a device or program capable of performing encoding and decoding on a digital data stream or signal. ... Pulse-code modulation (PCM) is a modulation technique. ...


G.722.1 offers lower bit-rate compressions. A more recent variant, G.722.2, also known as AMR-WB ("Adaptive Multirate Wideband"), offers even lower bit-rate compressions, as well as the ability to quickly adapt to varying compressions as the network topography mutates. In the latter case, bandwidth is automatically conserved when network congestion is high. When congestion returns to a normal level, a lower-compression, higher-quality bitrate is restored. G.722. ... Adaptive Multi Rate - WideBand or AMR-WB is a speech coding standard developed after the AMR using same technology like ACELP. The codec provides excellent speech quality due to wider speech bandwidth of 50 - 7000 Hz compared to narrowband speech codecs which in general are optimized for POTS wireline quality...


G.722 and its variants sample audio data at a rate of 16 kHz, four times that of traditional telephony interfaces, which results in superior audio quality and clarity.

Contents

Applications

G.722 is useful in fixed network voice over IP applications, where the required bandwidth (up to 64k) is typically not prohibitive, offering a significant improvement in speech quality over popular codecs such as G.711, without an excessive increase in implementation complexity. Environments where bandwidth is more constrained may prefer AMR-WB. An overview of how VoIP works A typical analog telephone adapter for connecting an ordinary phone to a VoIP network Ciscos implementation of VoIP - IP Phone Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of... G.711 is an ITU-T standard for audio companding. ...


RTP Encapsulation

G.722 VoIP is typically carried in RTP payload type 9 [2]. Note that IANA records the clock rate for type 9 G.722 as 8kHz (instead of 16kHz), RFC3551[3] clarifies that this is due to an historical error and is retained in order to maintain backward compatibility. Consequently correct implementations represent the value 8,000 where required but encode and decode audio at 16kHz. The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. ... The Internet Assigned Numbers Authority (IANA) is the entity that oversees global IP address allocation, DNS root zone management, and other Internet protocol assignments. ...


Licensing

G.722 patents have expired, so it is freely available.


See also

G.711 is an ITU-T standard for audio companding. ... G.722. ... Adaptive Multi Rate - WideBand or AMR-WB is a speech coding standard developed after the AMR using same technology like ACELP. The codec provides excellent speech quality due to wider speech bandwidth of 50 - 7000 Hz compared to narrowband speech codecs which in general are optimized for POTS wireline quality... This article is in need of attention from an expert on the subject. ... G.723. ... G.726 is ITU-T speech codec operating at bit rates of 16-40 kbit/s. ... G.728 is a ITU-T standard for speech coding operating at 16 kbit/s. ... G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. ... G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. ... The following is a list of codecs. ...

References

  1. ^ ITU-T G.722 page ITU-T Recommendation G.722 (11/88), "7 kHz audio-coding within 64 kbit/s"
  2. ^ IANA: Authoritative repository for RTP Parameters
  3. ^ RFC 3551Request For Comments 3551: RTP Profile for Audio and Video Conferences with Minimal Control. Schulzrinne & Casener, July 2003. Also Internet Standard 65.


 

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