By Colin Perkins
The Real-time shipping Protocol (RTP) presents a framework for supply of audio and video throughout IP networks with extraordinary caliber and reliability. In RTP: Audio and Video for the web, Colin Perkins, a pace-setter of the RTP standardization procedure within the IETF, bargains readers special technical tips for designing, enforcing, and handling any RTP-based process. by means of bringing jointly the most important info that was once formerly scattered or tough to discover, Perkins has created an important source that allows execs to leverage RTP's merits in quite a lot of Voice-over IP (VoIP) and streaming media purposes. He demonstrates how RTP helps audio/video transmission in IP networks, and stocks innovations for maximizing functionality, robustness, safety, and privateness. finished, really transparent, and replete with examples, this ebook is the definitive RTP reference for each audio/video program fashion designer, developer, researcher, and administrator. Key insurance contains: *RTP's ambitions, layout philosophy, and relationships with different protocols *The psychology of human notion within the layout of media supply structures *RTP info move and regulate protocols, together with framing, loss detection, reception caliber suggestions, and club keep an eye on *Media playout, timing, and synchronization, together with lip synchronization *Mitigating community difficulties: errors concealment, blunders correction, and congestion regulate *Optimizing functionality over low-speed hyperlinks: header compression, multiplexing, and tunneling *Integrating major media formats and criteria into RTP structures *Securing RTP classes: encryption, authentication, and the hot safe RTP profile for instant networks *Extensive references and sensible examples all through 0672322498B05092003
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The Real-time shipping Protocol (RTP) offers a framework for supply of audio and video throughout IP networks with unheard of caliber and reliability. In RTP: Audio and Video for the web, Colin Perkins, a pace-setter of the RTP standardization technique within the IETF, deals readers precise technical counsel for designing, enforcing, and handling any RTP-based process.
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Extra info for RTP: Audio and Video for the Internet
Although data packets are typically sent every few milliseconds, the control protocol operates on the scale of seconds. The information sent in RTCP is necessary for synchronization between media streams for example, for lip synchronization between audio and video and can be useful for adapting the transmission according to reception quality feedback, and for identifying the participants. The RTP control protocol is discussed further in Chapter 5. Part II: Media Transport Using RTP 37 38 Part II: Media Transport Using RTP RTP supports the notion of mixers and translators, middle boxes that can operate on the media as it flows between endpoints.
A secondary use of the sequence number is to allow reconstruction of the order in which packets were sent. A receiver does not necessarily care about this because many payload formats allow for decoding of packets in any order but sorting the packets into order as they are received may make loss detection easier. The design of playout buffer algorithms is discussed in more detail in Chapter 6, Media Capture, Playout, and Timing. TIMESTAMP The RTP timestamp denotes the sampling instant for the first octet of media data in a packet, and it is used to schedule playout of the media data.
The features that can be specified by RTP profiles include the following: • Mapping between the payload type identifier in the RTP header and the payload format specifications (which describe how individual media codecs are to be used with RTP). Each profile will reference multiple payload formats and may indicate how particular signaling protocols (for example, SDP15) are used to describe the mapping. • The size of the payload type identifier field in the RTP header, and the number of bits used to mark events of interest within a media stream.