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H.263 is a video compression standard originally designed as a low-bit-rate compressed format for videotelephony. It was standardized by the ITU-T Video Coding Experts Group (VCEG) in a project ending in 1995/1996. It is a member of the H.26x family of video coding standards in the domain of the ITU-T.
The Real-Time Streaming Protocol (RTSP) is an application-level network protocol designed for multiplexing and packetizing multimedia transport streams over a suitable transport protocol. RTSP is used in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between endpoints. Clients of media servers issue commands such as play, record and pause, to facilitate real-time control of the media streaming from the server to a client or from a client to the server.
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
The Session Description Protocol (SDP) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. Its predominant use is in support of streaming media applications, such as voice over IP (VoIP) and video conferencing. SDP does not deliver any media streams itself but is used between endpoints for negotiation of network metrics, media types, and other associated properties. The set of properties and parameters is called a session profile.
G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. It is an ITU-T standard (Recommendation) for audio encoding, titled Pulse code modulation (PCM) of voice frequencies released for use in 1972.
The RTP Control Protocol (RTCP) is a binary-encoded out-of-band signaling protocol that functions alongside the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in RFC 3550. RTCP provides statistics and control information for an RTP session. It partners with RTP in the delivery and packaging of multimedia data but does not transport any media data itself.
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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.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.
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MPEG transport stream or simply transport stream (TS) is a standard digital container format for transmission and storage of audio, video, and Program and System Information Protocol (PSIP) data. It is used in broadcast systems such as DVB, ATSC and IPTV.
Real-Time Messaging Protocol (RTMP) is a communication protocol for streaming audio, video, and data over the Internet. Originally developed as a proprietary protocol by Macromedia for streaming between Flash Player and the Flash Communication Server, Adobe has released an incomplete version of the specification of the protocol for public use.
Internet Low Bitrate Codec (iLBC) is a royalty-free narrowband speech audio coding format and an open-source reference implementation (codec), developed by Global IP Solutions (GIPS) formerly Global IP Sound. It was formerly freeware with limitations on commercial use, but since 2011 it is available under a free software/open source license as a part of the open source WebRTC project. 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, usually the Real-time Transport Protocol (RTP).
ATSC-M/H is a U.S. standard for mobile digital TV that allows TV broadcasts to be received by mobile devices.
Pulse-code modulation (PCM) is a method used to digitally represent analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.
RTP-MIDI is a protocol to transport MIDI messages within Real-time Transport Protocol (RTP) packets over Ethernet and WiFi networks. It is completely open and free, and is compatible both with LAN and WAN application fields. Compared to MIDI 1.0, RTP-MIDI includes new features like session management, device synchronization and detection of lost packets, with automatic regeneration of lost data. RTP-MIDI is compatible with real-time applications, and supports sample-accurate synchronization for each MIDI message.
AES67 is a technical standard for audio over IP and audio over Ethernet (AoE) interoperability. The standard was developed by the Audio Engineering Society and first published in September 2013. It is a layer 3 protocol suite based on existing standards and is designed to allow interoperability between various IP-based audio networking systems such as RAVENNA, Wheatnet, Livewire, Q-LAN and Dante.
JPEG XS is an interoperable, visually lossless, low-latency and lightweight image and video coding system used in professional applications. Applications of the standard include streaming high-quality content for virtual reality, drones, autonomous vehicles using cameras, gaming, and broadcasting. It was the first ISO codec ever designed for this specific purpose. JPEG XS, built on core technology from both intoPIX and Fraunhofer IIS, is formally standardized as ISO/IEC 21122 by the Joint Photographic Experts Group with the first edition published in 2019. Although not official, the XS acronym was chosen to highlight the eXtra Small and eXtra Speed characteristics of the codec. Today, the JPEG committee is still actively working on further improvements to XS, with the second edition scheduled for publication and initial efforts being launched towards a third edition.
SMPTE 2110 is a suite of standards from the Society of Motion Picture and Television Engineers (SMPTE) that describes how to send digital media over an IP network.
References
- ↑ RFC 3551, RTP Profile for Audio and Video Conferences with Minimal Control, H. Schulzrinne, S. Casner, The Internet Society (July 2003).
- ↑ "RFC 2586 - The Audio/L16 MIME content type". May 1999. Retrieved 2010-03-16.
- ↑ "RFC 3108 - Conventions for the use of the Session Description Protocol (SDP) for ATM Bearer Connections". May 2001. Retrieved 2010-03-16.
- ↑ "RFC 4856 - Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences - Registration of Media Type audio/L16". March 2007. Retrieved 2010-03-16.
- ↑ XIL Programmer's Guide, Chapter 22 "CellB Codec". August 1997. Retrieved on 2014-07-19.
- ↑ nv - network video on Henning Schulzrinne's website, Network Video on The University of Toronto's website, Retrieved on 2009-07-09.
- ↑ Ron Frederick Github with source code