 | |
| Developer(s) | 64 Characters |
|---|---|
| Stable release | 1.4.2 / April 3, 2020 |
| Written in | Objective-C, Swift |
| Operating system | macOS |
| Available in | English, German, Russian |
| License | GPL |
| Website | www.64characters.com/telephone |
Telephone is a softphone for macOS developed by 64 Characters. It uses Session Initiation Protocol for communication. Telephone is distributed as free software under GPL.
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Bandwidth is the difference between the upper and lower frequencies in a continuous band of frequencies. It is typically measured in hertz, and depending on context, may specifically refer to passband bandwidth or baseband bandwidth. Passband bandwidth is the difference between the upper and lower cutoff frequencies of, for example, a band-pass filter, a communication channel, or a signal spectrum. Baseband bandwidth applies to a low-pass filter or baseband signal; the bandwidth is equal to its upper cutoff frequency.
Digital enhanced cordless telecommunications , usually known by the acronym DECT, is a standard primarily used for creating cordless telephone systems. It originated in Europe, where it is the universal standard, replacing earlier cordless phone standards, such as 900 MHz CT1 and CT2.
Speex is an audio compression codec specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on VoIP applications and podcasts. It is based on the CELP speech coding algorithm. Speex claims to be free of any patent restrictions and is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/RTP. It may also be used with the FLV container format.
In communications, a system is wideband when the message bandwidth significantly exceeds the coherence bandwidth of the channel. Some communication links have such a high data rate that they are forced to use a wide bandwidth; other links may have relatively low data rates, but deliberately use a wider bandwidth than "necessary" for that data rate in order to gain other advantages; see spread spectrum.
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), and was introduced in 1996. The wide-band extension of G.729 is called G.729.1, which equals G.729 Annex J.
G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on the same technology is G.726.
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. It uses a modified discrete cosine transform audio data compression algorithm.
A discone antenna is a version of a biconical antenna in which one of the cones is replaced by a disc. It is usually mounted vertically, with the disc at the top and the cone beneath.
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 (PCM) and G.728 (CELP), and discrete cosine transform (DCT) video codecs such as H.261 and H.263.
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.
internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS). It is suitable for VoIP applications and streaming audio. The encoded blocks have to be encapsulated in a suitable protocol for transport, e.g. RTP.
TIMIT is a corpus of phonemically and lexically transcribed speech of American English speakers of different sexes and dialects. Each transcribed element has been delineated in time.
FreeSWITCH is a free and open-source application server for real-time communication, WebRTC, telecommunications, video and Voice over Internet Protocol (VoIP). Multiplatform, it runs on Linux, Windows, macOS and FreeBSD. It is used to build PBX systems, IVR services, videoconferencing with chat and screen sharing, wholesale least-cost routing, Session Border Controller (SBC) and embedded communication appliances. It has full support for encryption, ZRTP, DTLS, SIPS. It can act as a gateway between PSTN, SIP, WebRTC, and many other communication protocols. Its core library, libfreeswitch, can be embedded into other projects. It is licensed under the Mozilla Public License (MPL), a free software license.
TIA/EIA-920 is a telecommunications standard from the US Telecommunications Industry Association, a 1988 offshoot of the EIA. The standard establishes wideband audio performance requirements for wireline telephones which transmit their signals digitally. Audio wideband is defined as 150 Hz to 6800 Hz. The technical standard refers to handsets, headsets, and speakerphones. It does not specify digital protocols or encoding formats. Nor does it concern how the transducers are connected to the telephone.
Linphone is a free voice over IP softphone, SIP client and service. It may be used for audio and video direct calls and calls through any VoIP softswitch or IP-PBX. Also Linphone provides the possibility to exchange instant messages. It has a simple multilanguage interface based on GTK+ for GUI and can also be run as a console-mode application on Linux.
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.
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.