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ipDTL is an IP codec that runs in a web browser, used for remote broadcasts in television, radio, and voice-over. [1] It serves as a replacement for and is compatible with older ISDN audio codecs. Developed by former BBC sound engineer Kevin Leach, ipDTL uses the open-source codec OPUS and operates within web browsers, specifically Blink-based browsers like Google Chrome or Opera. [2]
ipDTL was invented by former BBC sound engineer Kevin Leach and uses the open-source codec OPUS since it first became available in the Google Chrome web browser. [3] It enables higher audio quality than ISDN by accessing a website through a web browser. [1] Audio quality of 72 kbit/s mono for voice contributors, [4] 320 kbit/s for outside broadcasts with music, and 3 Mbit/s video at 1080p for video contributions on TV programs are possible. [1] This technology was launched in 2013. [5]
ipDTL uses WebRTC and Web audio technologies. It is designed primarily for Blink-based browsers like Google Chrome or Opera and runs on all platforms except iOS where these browsers are supported. The codecs used are Opus for audio and VP8 for video. The supported audio bandwidth is up to 320 kbit/s (stereo) and up to 3 Mbit/s for video (1080p). [1]
Connections are established point-to-point and have DTLS encryption. [3] Where a point-to-point connection is not possible, TURN relay servers are used to route the audio. TURN servers in the US, UK, Brazil, Australia, and Japan are available, with an independent backup system being maintained at ipdtl2.com. Connections can also be made through a special URL that allows users to access another account and connect with it. [1] ipDTL uses a proprietary signaling method but also supports SIP for interoperability with other devices and applications such as Comrex Access and Media5-fone, and can transcode between Opus, G.722 and G.711. [6] It also supports interoperability with legacy ISDN hardware via cloud-based bridging servers. [7]
ipDTL powers hybrIP – a talk show system, which allows screening calls using any computer.
Integrated Services Digital Network (ISDN) is a set of communication standards for simultaneous digital transmission of voice, video, data, and other network services over the digitalised circuits of the public switched telephone network. Work on the standard began in 1980 at Bell Labs and was formally standardized in 1988 in the CCITT "Red Book". By the time the standard was released, newer networking systems with much greater speeds were available, and ISDN saw relatively little uptake in the wider market. One estimate suggests ISDN use peaked at a worldwide total of 25 million subscribers at a time when 1.3 billion analog lines were in use. ISDN has largely been replaced with digital subscriber line (DSL) systems of much higher performance.
Speech coding is an application of data compression to 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.
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 voice over IP applications and podcasts. It is based on the code excited linear prediction speech coding algorithm. Its creators claim Speex to be free of any patent restrictions and it 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 telecommunications, broadband or high speed is the wide-bandwidth data transmission that exploits signals at a wide spread of frequencies or several different simultaneous frequencies, and is used in fast Internet access. The transmission medium can be coaxial cable, optical fiber, wireless Internet (radio), twisted pair cable, or satellite.
Motion JPEG is a video compression format in which each video frame or interlaced field of a digital video sequence is compressed separately as a JPEG image.
In telecommunications and computing, bit rate is the number of bits that are conveyed or processed per unit of time.
Adaptive Multi-Rate Wideband (AMR-WB) is a patented wideband speech audio coding standard developed based on Adaptive Multi-Rate encoding, using a similar methodology to 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.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.
Videotelephony is the two-way or multipoint reception and transmission of audio and video signals by people in different locations for real-time communication. A videophone is a telephone with a video camera and video display, capable of simultaneous video and audio communication. Videoconferencing implies the use of this technology for a group or organizational meeting rather than for individuals, in a videoconference. Telepresence may refer either to a high-quality videotelephony system or to meetup technology, which can go beyond video into robotics. Videoconferencing has also been called visual collaboration and is a type of groupware.
Blackbird is an integrated internet video platform, video editing software, covering non-linear editing and publishing for broadcast, web and mobile.
The MPEG-4 Low Delay Audio Coder is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) standard. It was published in MPEG-4 Audio Version 2 and in its later revisions.
Audio over IP (AoIP) is the distribution of digital audio across an IP network such as the Internet. It is used increasingly to provide high-quality audio feeds over long distances. The application is also known as audio contribution over IP (ACIP) in reference to the programming contributions made by field reporters and remote events. Audio quality and latency are key issues for contribution links. In the past, these links have made use of ISDN services but these have become increasingly difficult or expensive to obtain.
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 100 Hz to 17 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. 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.
aptX is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications.
IP codecs are used to send video or audio signals over an IP network such as the Internet. The initials "IP" here stand for "Internet Protocol", while the term "codec" is short for "encoder/decoder" or "compressor/decompressor".
The HTML5 specification introduced the video element for the purpose of playing videos, partially replacing the object element. HTML5 video is intended by its creators to become the new standard way to show video on the web, instead of the previous de facto standard of using the proprietary Adobe Flash plugin, though early adoption was hampered by lack of agreement as to which video coding formats and audio coding formats should be supported in web browsers. As of 2020, HTML5 video is the only widely supported video playback technology in modern browsers, with the Flash plugin being phased out.
WebRTC is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) via application programming interfaces (APIs). It allows audio and video communication and streaming to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps.
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.
Web Call Server is unified intermedia server software developed by Flashphoner. It is a server-side platform, implemented in Java, dedicated for streaming video over wide range of communication protocols, including: