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RTP Payload Formats
The Real-time Transport Protocol (RTP) specifies a general-purpose data format and network protocol for transmitting digital media streams on Internet Protocol (IP) networks. The details of media encoding, such as signal sampling rate, frame size and timing, are specified in an RTP payload format. The format parameters of the RTP payload are typically communicated between transmission endpoints with the Session Description Protocol (SDP), but other protocols, such as the Extensible Messaging and Presence Protocol (XMPP) may be used. Audio and video payload types RFC 3551, entitled RTP Profile for Audio and Video (RTP/AVP), specifies the technical parameters of payload formats for audio and video streams. The standard also describes the process of registering new payload types with IANA; additional payload formats and payload types are defined in the following specifications: * , Standard 65, ''RTP Profile for Audio and Video Conferences with Minimal Control'' * , ''Media Type Regis ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Real-time Transport Protocol
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. RTP typically runs over User Datagram Protocol (UDP). RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
GSM-FR
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system. It uses linear predictive coding (LPC). The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample (often padded out to 33 bytes/20 ms or 13.2 kbit/s). The quality of the coded speech is quite poor by modern standards, but at the time of development (early 1990s) it was a good compromise between computational complexity and quality, requiring only on the order of a million additions and multiplications per second. The codec is still widely used in networks around the world. Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate. Technology ''GSM-FR'' is specified in ETSI 06.10 (ETS 300 961) and is based on RPE-LTP ( Regular Pulse Excitation - Long Term Prediction) speech coding paradigm. Like many other linear predicti ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sun Microsystems
Sun Microsystems, Inc. (Sun for short) was an American technology company that sold computers, computer components, software, and information technology services and created the Java programming language, the Solaris operating system, ZFS, the Network File System (NFS), and SPARC microprocessors. Sun contributed significantly to the evolution of several key computing technologies, among them Unix, RISC processors, thin client computing, and virtualized computing. Notable Sun acquisitions include Cray Business Systems Division, Storagetek, and ''Innotek GmbH'', creators of VirtualBox. Sun was founded on February 24, 1982. At its height, the Sun headquarters were in Santa Clara, California (part of Silicon Valley), on the former west campus of the Agnews Developmental Center. Sun products included computer servers and workstations built on its own RISC-based SPARC processor architecture, as well as on x86-based AMD Opteron and Intel Xeon processors. Sun also develope ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Comfort Noise
Comfort noise (or comfort tone) is synthetic background noise used in radio and wireless communications to fill the artificial silence in a transmission resulting from voice activity detection or from the audio clarity of modern digital lines. Some modern telephone systems (such as wireless and VoIP) use voice activity detection (VAD), a form of squelching where low volume levels are ignored by the transmitting device. In digital audio transmissions, this saves bandwidth of the communications channel by transmitting nothing when the source volume is under a certain threshold, leaving only louder sounds (such as the speaker's voice) to be sent. However, improvements in background noise reduction technologies can occasionally result in the complete removal of all noise. Although maximizing call quality is of primary importance, exhaustive removal of noise may not properly simulate the typical behavior of terminals on the PSTN system. The result of receiving total silence, es ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-2
MPEG-2 (a.k.a. H.222/H.262 as was defined by the ITU) is a standard for "the generic coding of moving pictures and associated audio information". It describes a combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and H.265/HEVC, backwards compatibility with existing hardware and software means it is still widely used, for example in over-the-air digital television broadcasting and in the DVD-Video standard. Main characteristics MPEG-2 is widely used as the format of digital television signals that are broadcast by terrestrial (over-the-air), cable, and direct broadcast satellite TV systems. It also specifies the format of movies and other programs that are distributed on DVD and similar discs. TV stations, TV receivers, DVD players, and other equipment are ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-1
MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s (26:1 and 6:1 compression ratios respectively) without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical. Today, MPEG-1 has become the most widely compatible lossy audio/video format in the world, and is used in a large number of products and technologies. Perhaps the best-known part of the MPEG-1 standard is the first version of the MP3 audio format it introduced. The MPEG-1 standard is published as ISO/IEC 11172 – Information technology—Coding of moving pictures and associated audio for digital storage media at up to about 1.5 Mbit/s. The standard consists of the following five ''Parts'': #Systems (storage and synchronization of video, audio, and other data together) #Video (compressed video content) #Audio (compressed audio content) #Conformance ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
RTAudio
RTAudio is a Microsoft produced adaptive wide-band speech codec. It is used by Microsoft Office Communications Server (OCS) and the related OCS clients ( Microsoft Office Communicator, and Microsoft Live Meeting Console). RTAudio was designed for real-time two-way Voice over IP (VoIP) applications. Some of the target applications include games, audio conferencing, and wireless applications over IP. RTAudio is the preferred Microsoft Real-Time audio codec, and is the default voice codec for Microsoft Microsoft Corporation is an American multinational technology corporation producing computer software, consumer electronics, personal computers, and related services headquartered at the Microsoft Redmond campus located in Redmond, Washi ...’s Unified Communications platforms. The RTAudio encoder is capable of encoding single-channel (mono), 16 bit per sample audio signals. The encoder can be configured to operate either in the Narrow Band mode (8 kHz sampling rate ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Siren (codec)
Siren is a family of patented, transform-based, wideband audio coding formats and their audio codec implementations developed and licensed by PictureTel Corporation (acquired by Polycom, Inc. in 2001). There are three Siren codecs: Siren 7, Siren 14 and Siren 22. Editions Siren 7 (or Siren7 or simply Siren) provides 7 kHz audio, bit rates 16, 24, 32 kbit/s and sampling frequency 16 kHz. Siren is derived from PictureTel's PT716plus algorithm. In 1999, ITU-T approved G.722.1 recommendation, which is based on Siren 7 algorithm. It was approved after a four-year selection process involving extensive testing. G.722.1 provides only bit rates 24 and 32 kbit/s and does not support Siren 7's bit rate 16 kbit/s. The algorithm of Siren 7 is identical to its successor, G.722.1, although the data formats are slightly different. Siren 14 (or Siren14) provides 14 kHz audio, bit rates 24, 32, 48 kbit/s for mono, 48, 64, 96 kbit/s for stereo and sampling frequency 32 kHz. Sir ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
GSM 06
The Global System for Mobile Communications (GSM) is a standard developed by the European Telecommunications Standards Institute (ETSI) to describe the protocols for second-generation ( 2G) digital cellular networks used by mobile devices such as mobile phones and tablets. GSM is also a trade mark owned by the GSM Association. GSM may also refer to the Full Rate voice codec. It was first implemented in Finland in December 1991. By the mid-2010s, it became a global standard for mobile communications achieving over 90% market share, and operating in over 193 countries and territories. 2G networks developed as a replacement for first generation ( 1G) analog cellular networks. The GSM standard originally described a digital, circuit-switched network optimized for full duplex voice telephony. This expanded over time to include data communications, first by circuit-switched transport, then by packet data transport via General Packet Radio Service (GPRS), and Enhanced Data Rates for G ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Comfort Noise
Comfort noise (or comfort tone) is synthetic background noise used in radio and wireless communications to fill the artificial silence in a transmission resulting from voice activity detection or from the audio clarity of modern digital lines. Some modern telephone systems (such as wireless and VoIP) use voice activity detection (VAD), a form of squelching where low volume levels are ignored by the transmitting device. In digital audio transmissions, this saves bandwidth of the communications channel by transmitting nothing when the source volume is under a certain threshold, leaving only louder sounds (such as the speaker's voice) to be sent. However, improvements in background noise reduction technologies can occasionally result in the complete removal of all noise. Although maximizing call quality is of primary importance, exhaustive removal of noise may not properly simulate the typical behavior of terminals on the PSTN system. The result of receiving total silence, es ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
QCELP
Qualcomm code-excited linear prediction (QCELP), also known as Qualcomm PureVoice, is a speech codec developed in 1994 by Qualcomm to increase the speech quality of the IS-96A codec earlier used in CDMA networks. It was later replaced with EVRC since it provides better speech quality with fewer bits. The two versions, ''QCELP8'' and ''QCELP13'', operate at 8 and 13 kilobits per second (Kbit/s) respectively. In CDMA systems, a QCELP vocoder A vocoder (, a portmanteau of ''voice'' and ''encoder'') is a category of speech coding that analyzes and synthesizes the human voice signal for audio data compression, multiplexing, voice encryption or voice transformation. The vocoder ... converts a sound signal into a signal transmissible within a circuit. In wired systems, voice signals are generally sampled at 8 kHz (that is, 8,000 sample values per second) and then encoded by 8-bit quantization for each sample value. Such a system transmits at 64 kbit/s, an expensive ra ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Linear PCM
Pulse-code modulation (PCM) is a method used to digitally represent sampled 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 regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform. This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm). Though ''PCM'' is a more general term, it is often used to describe data encoded as LPCM. A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
