Streaming Media

Unicast Redundant traffic Multicast One to many

Video Multicast Multicast Support multicast enabled network Real-time Requirements Support

Supporting Real-time Requirements QoS and Resource reservation Resource Reservation to bound data delivery delay, loss, jitter etc. Adaptive Rate Control Adjust video traffic characteristics to suit the Internet.

Multicast Group Addressing Distribution Tree Join(graft) Leave(prune)

Video Traffic Periodic generation of frames at regular intervals. Variable bit rate. Frame periodicity must be maintained for the video to appear “smooth” Data unavailable at playout is useless. Jitter (variability in interarrival times)

Buffering and Start-up Latency Congestion leads to Data Loss Decreasing Data Rate Error Control Summary Delay Sensitive, Loss Insensitive

Multicast and Heterogeneity The Internet is Heterogeneous Infra-structural (Spatial) Traffic density (Temporal) Administrative Fairness Goal Every receiver should receive video that is commensurate to the resources available. Is this fair to “other” traffic?

Fairness Intra-session fairness Inter-session fairness

Rate Adaptive Digital Video Compression, Scene Complexity, Motion Video Encoder Smoothing Buffer Feedback Network

Raw video stream is fed to encoder Encoder sends encoded data to buffer Buffer level provides feedback Feedback is used to control data output rate at encoder. Quantization, frame rate, pixel resolution etc. are controlled.

Network feedback can also be used. Queueing information (internal to the network) End-system information

Adaptive Bit-Rate Video Single Stream Adaptive Approach Replicated Streams Adaptive Approach Layered Video Streams Approach

RTP Real-Time Transport Protocol End-to-end Protocol NO notion of “Connection”. (hence unreliable) Application level Requires framing and segmentation be taken care of by lower layers.

RTP (continued) Divided into two parts (consecutive ports for UDP) Data (audio + video packets, even- numbered port) Control Can use single PDU in case UDP is not used.

Real-time Transport Protocol

RTP Data Packets 12 byte header data (video/audio) can be encapsulated in encoding-specific layer.

RTP Data Packet Header Payload Type (1 byte) eg: JPEG etc. Timestamp (32 bits) generation instant of the data Sequence marker (16 bits) packet seq. number to help loss detection Marker bit end of frame for video beginning of talk-spurt for audio

RTP Data Packet Header (contd) Synchronization Source Identifier (32 bits) randomly generated identifying session source.

RTP Control Channel Control protocol called RTCP QoS monitoring and Congestion Control multicast all other receivers know how others are doing sender-report, helps receivers compute data-rate Intermedia Synchronization wall-clock time + RTP timestamp allows synch of audio and video Identification Detailed identification of participant instead of just a 32 bit identifier. Session size estimation and scaling scaled to 5% of data rate

RTCP Packets Several types to carry a variety of information Source description (SDES) CNAME, , location, name,... Sender report (SR) Bytes sent -> estimated rate Timestamp -> synchronization Receiver report (RR) Loss rate, interarrival jitter, roundtrip delay Explicit leave (BYE) Compound packets (SDES CNAME + RR)

RTCP traffic Control RTP session scale: two to thousands of participants RTCP traffic increases with session size Want to keep to small fraction of data bandwidth (5%) Randomized response with rate decreasing as number of participants increases Give active senders more bandwidth But limited by tolerable age of status

Single Stream Video Multicast Adjust video output rate Three parameters refresh rate (?) quantizer (color scheme 4:2:2, 4:1:1….) movement detection threshold (what defines motion) Application can specify which of these to control

Single Stream Video Multicast RTCP is used for feedback Feedback implosion probabilistic probing Fair? (No…..)

Replicated Streams Destination Set Grouping Multiple replicated streams on different multicast addresses. Different quality and data rates. Receivers can switch streams

Switching Streams Congestion due to presence of two streams simultaneously on the same link Bandwidth Control Protocol Congestion History Checking before stream switch. Local Area Bandwidth Limit restricts the number of streams received in local area.

Layered Video Multicast Disjoint layers on different addresses Cumulative subscription Many protocols making different assumptions

RLM Receiver based Sender does not participate Receivers share loss information Receivers join and drop groups based on these shared loss reports. Receivers back off when they or other receivers see congestion. The higher the layer, the longer the back-off duration.

Problems with RLM Receiver Consensus Fast Leaves and Joins Impact of failed experiments on topologically unrelated receivers. UNFAIR Arguably the most cited and most maligned protocol!!

TCP rate-based Congestion Control Analyze TCP to come with a magic formula to describe Bandwidth = 1.3 * MTU / (RTT * sqrt(Loss)) Adapt sender rate to match such a formula. But what is RTT? Let receivers make this decision. Define loss thresholds based on this formula, for each layer. If loss exceeds this threshold, drop a layer…

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Summary Multicast RTP/RTCP Rate Adaptation Issues: Fairness Intra-session Inter-session Stability Deployability Administrative Issues