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Transmission of frame 3 may be blocked by TCP.
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Figure 6.2 Retransmission of lost packet under TCP may cause media data to miss playback deadline at the client
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until the congestion window grows back to normal after receiving a number of acknowledgements from the receiver. Again, this sender throttling will cause problems in media streaming as deferring transmission of the media data may cause them to miss the playback deadline, thus rendering them useless even if they are eventually received by the client. The User Datagram Protocol (UDP), on the other hand, does not suffer from the problems of TCP as it is a relatively simple protocol that transfers datagrams without ow control, congestion control, or any error control at all. Therefore, the protocol itself will not introduce additional delay (ignoring processing time and packetization delay) like the ow control and congestion control algorithm in TCP, making it suitable for delivering time-sensitive media data. Nevertheless, in media streaming it is sometimes still necessary to perform ow control, to react to network congestion, as well as handling packet losses. The key is that when performing these functions the timing and bandwidth requirement of the media data must be taken into account. This can be achieved by implementing another layer of streaming protocol on top of UDP, where the streaming protocol will handle the streaming-speci c functions while UDP is simply used to deliver the data and control messages. We review in the next section some of the more popular streaming protocols in the Internet.
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6.2 Specialized Streaming Protocols
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Over the years a number of streaming protocols have been developed both by commercial companies and the Internet community. On the commercial side, streaming solution companies often develop their own proprietary streaming protocols for use in their streaming products. For example, Microsoft developed a Microsoft Media Services (MMS) for use in its Windows
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Media streaming solution. MMS employs TCP for the exchange of control messages and can send the media data over either UDP or TCP. RealNetworks also developed their own RealNetworks Data Transport (RDT) for use in their streaming solution. Because of the proprietary nature of these protocols we will not cover them further in this chapter. On the other hand, the Internet community has also developed open standards for media streaming. This includes the Real Time Streaming Protocol (RTSP) de ned in RFC 2326 [7], the Real-time Transport Protocol (RTP) and the RTP Control Protocol (RTCP), rst introduced in RFC 1889, later revised in RFC 3550 [8], which became an of cial standard in May 2004.
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6.2.1 Real-Time Streaming Protocol (RTSP)
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The Real-Time Streaming Protocol (RTSP) is an application-layer protocol designed to control the delivery of media data (e.g., play, pause, and seek) with embedded timing information, such as audio and video. The protocol is independent of the lower-layer protocol. Thus, RTSP can be carried over TCP, UDP, or other transport protocols. The syntax of RTSP shares many similarities with HTTP/1.1, thus simplifying implementation and deployment. However, besides the syntax similarities, RTSP differs from HTTP in many important ways. First, unlike HTTP, RTSP is a stateful protocol, thus requiring the host to maintain state information of a streaming session across multiple RTSP requests. Second, both the RTSP server and client can issues RTSP requests. Finally, the media data are to be delivered out-ofband, i.e., using a separate protocol such as, but not limited to, the Real-time Transport Protocol. In a typical streaming application (see Figure 6.3), the client will rst obtain a presentation description le using out-of-band methods (e.g., through the web using HTTP). The presentation description le describes one or more presentations, each composed of one or more synchronized media streams. The presentation description le also contains properties of the
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