2Characteristics of Multimedia Data Voluminous — they demand very high data rates, possibly dozens or hundreds of Mbps.Real-time and interactive — they demand low delay and synchronization between audio and video for “lip sync”. In addition, applications such as video conferencing and interactive multimedia also require two-way traffic.Sometimes bursty — data rates fluctuate drastically, e.g., no traffic most of the time but burst to high volume in video-on-demand.
3Quality of Multimedia Data Transmission Quality of Service (QoS) depends on many parameters:Data rate: a measure of transmission speed.Latency (maximum frame/packet delay): maximum time needed from transmission to reception.Packet loss or error: a measure (in percentage) of error rate of the packetized data transmission.Jitter: a measure of smoothness of the audio/video playback, related to the variance of frame/packet delays.
4Fig. 1: Jitters in frame playbacks. (a) High jitter, (b) Low jitter.
5Multimedia Service Classes Real-Time (also Conversational): two-way traffic, low latency and jitter, possibly with prioritized delivery, e.g., voice telephony and video telephony.Priority Data: two-way traffic, low loss and low latency, with prioritized delivery, e.g., E-commerce applications.Silver: moderate latency and jitter. One-way traffic, e.g., streaming video, or two-way traffic (also Interactive), e.g., web surfing, Internet games.Best Effort (also Background): no real-time requirement, e.g., downloading or transferring large files (movies).Bronze: no guarantees for transmission.
6Table 1: Requirement on Network Bandwidth / Bit-rate
7Table 2: Tolerance of Latency and Jitter in Digital Audio and Video
8What is Steaming Video Download mode: No delay bound Streaming mode: delay bound
9Challenges for Quality Video Transport Time-Varying Available BandwidthNo bandwidth reservation
12Architecture for Video Streaming Upon the request, a streaming server retrieves compressed video/audio data from storage devies, then the application –layer QoS control module adapts the vide/audio bit steams according to the network status and QoS requirments. After this adaptation, the tranport protocol packetize the compressed bit streams and send the video/audio packet to the interent or wireless IP networks.Packets may be dropped or experince excesssive delay on the internet due to congestion; On wireless iP segments, packets may be damaged by bit errors. To improve the quality of video/audio transmission, continous media distribution services are deployed in the interenet. Packet are sucessive delived to reciever first pass through the transprot layers and then are processed by application layer before being decoded at the video/audo decoder. To achieve sychronization between video and audio presentations, media synchronization mechanism are required. From Figure 15.1, it can be seens that seven areas are colosely related, and that they are coherent consituents of the video streaming architecture.
13Video Compression Layered Video encoding/decoding D denotes the decoder
15Application-Layer QoS Control Congestion control (using rate control)Source-based, requiresRate-adaptive compression orRate shapingReceiver-basedHybridError ControlForward error correction (FEC)RetransmissionError resilient compressionError concealmentApplication layer QoS control maximizes video quality in the presence of pakcet loss and changes in available bandwidth. Application-layer QoS control techniques include congestion control and error control
16Congest Control Source-based Rate Control Rate shaper is a technique trough which the rate of precompressed video bit streams can be adapted to a target rate contraints.
17Mode-based ApproachThe source is reponsible for adapting video transmission rate. Typically, feedback is employed by source-based rate control mechanism. Based on the feedback inforamtion about the network, the sender can regulate the rate of video stream. The model based is based on a throughput model of a tcp connection. Specifically, the throughput of a TCP connection can be characterized by the following formula. The equaition is used to determine the sending rate of video stream. In tthis way, the video connection can avoid the congestionWhere is the throughput of a TCP connection, MTU (maximum transmission unit) is the packet size used by the connection, RTT is the round-trip time for the connection, and p is the packet-loss ratio experienced by the connection.
18Receiver-based Rate Control IP multicast for layered videoUnder receive-based rate control, recievers regulate the receivng rate of video streams by adding and dropping channels, whereas the sender does not particiatpe in rate control. Typically, reciever-based rate control is only applied to laeyred multicast
21Delay-constrained Retransmission When the receiver detects the loss of packet N:IfSend the request for packet N to the senderThe tolearance of error in estimaing RTT, the sender’s response time, the reciever
22Streaming Server Different from a web server Timing constraintsVideo-cassette-recorder(VCR) functions (e.g., fast forward/backward, random access, and pause/resume).Design of streaming serversReal-time operating systemSpecial disk scheduling schemes
23Media Synchronization Why media synchronizationExample: lip-synchronization (video/audio)
24Protocol for Streaming Video Network-layer protocol: Internet Protocol (IP)Transport protocolLower layer: UDP & TCPUpper layer: Real-time Transport Protocol (RTP) & Real-Time Control Protocol (RTCP)Session control protocolReal-time streaming protocol (RTSP): RealPlayerSession Initiation Protocol (SIP) : Microsoft Windows MediaPlayer; Internet Telephony
26Summary Challenges for Quality video transport Time-varying available bandwidthTime-varying delayPacket lossAn architecture for video streamingApplication-layer QoS controlStreaming serverMedia synchronization mechanismsProtocols for steaming media