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Computer Networks: Multimedia Applications

Computer Networks: Multimedia Applications. Ivan Marsic Rutgers University. Chapter 3 – Multimedia & Real-time Applications. Multimedia & Real-time Applications. Chapter 3. Topic : Traffic Sources & Models.  Source Coding Traffic Types Traffic Models Birth and Death Processes.

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Computer Networks: Multimedia Applications

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  1. Computer Networks:Multimedia Applications Ivan Marsic Rutgers University Chapter 3 – Multimedia & Real-time Applications

  2. Multimedia & Real-time Applications Chapter 3

  3. Topic:Traffic Sources & Models  Source Coding Traffic Types Traffic Models Birth and Death Processes

  4. Source Coding vs. Channel Coding Source Coding Decoding Channel Coding

  5. Signal Digitalization Source Coding – a simple example

  6. Speech Signal Digitalization Source Coding – also involves data compression (may be lossy)

  7. What is a Traffic Source

  8. Traffic Model for Voice Source Idle Call Talk

  9. Birth and Death Processes

  10. Topic:Delayed Playout for Jitter Control  Delay  Delay Variation (Jitter)  Jitter Buffer

  11. How Delay Jitter Arises

  12. Solution: Delayed Playout

  13. Problem & Tradeoff to Make • How to set the playout delay value? • Problem: Network delays change over time even for the same endpoints • Tradeoff: • Prefer to set the playout delay as small as possible for real-time applications (telephony) because of human psychological characteristics • But, small playout delay may cause too many packets to miss their playout deadline • Solution: Adaptive playout delay

  14. Recall from Chapter 2:TCP Retransmission Timer Calculation

  15. Adaptive Playout Problem: Cannot change playout delay during the speech Solution: Change playout delay during the silence periods … speech … silence … speech … silence … speech … i+1 change playout delay qi+1 i Play w/ playout delay qi estimate playout delay qi+1

  16. Topic:Multicast Routing  Reverse Path Forwarding (RPF) Algorithm  Spanning Tree Algorithms

  17. Multicast Routing

  18. Superimposed Shortest Paths = Tree

  19. Multicast Example 3.1

  20. Multicast – Solution

  21. Multicast Example

  22. Multicast: CBT Spanning Tree

  23. Data Packet Forwarding in CBT

  24. Multicast: Spanning Tree Teardown

  25. Topic:Differentiated Services  DiffServ Architecture

  26. DiffServ Architecture

  27. Topic:Multimedia Protocols  Real-time Transport Protocol (RTP) RTP Control Protocol (RTCP) SIP, SDP, …

  28. RTP: Real-time Transport Protocol TCP/IP protocol suite: RTP UDP Layer 3: Transport Layer 2: Internet IP Layer 1: Link

  29. RTP Header

  30. RTP Header (2) P = padding bit indicates if packet is padded to a required size (e.g., for encryption) X = extension bit indicates an extension header; rarely used b/c appl. defines own hdr CSRC count = # contributing source IDs in the header, if any M = marks packets w/ significant events Payload type = payload format indicates encoding scheme used for audio, video, etc.

  31. RTP Header (3) Sequence number = used by receiver for delay jitter removal Timestamp = used with seq. num. to detect pkt loss. Also to synchronize packets from different sources. Represents the sampling (creation) time of the 1st byte. Possible that successive packets have the same timestamp. E.g., a single video frame is transmitted in multiple RTP packets. SSRC identifier = unique synchronization source ID randomly chosen. Same for all packets from the same src/device. Enables receiver to group packets for playback. CSRC identifiers = list of contributing sources for the packet payload. Used when a mixer combines several streams of packets. Allows the receiver to identify the original senders.

  32. RTCP Header Format Common sender/receiver REPORT message header: All report messages have the same 8 byte header » version number (same as RTP) » padding indicator » reception report count (5 bits) » RTCP message type (8 bits) » RTCP message length (16 bits) » SSRC for the sender of this report (32 bits)

  33. Compound RTCP Packet Format

  34. Reception Report Blocks Each sender and receiver report should contain a reception report block for each synchronization source heard from since the last RTCP report Contents: source identifier for the block (SSRC) fraction of RTP packets from this source lost since the last report cumulative number of lost packets extended highest sequence number received estimated average RTP packet interarrival time jitter last SR timestamp received from this source delay since receiving the last SR report from this source

  35. RTCP Receiver Report Format

  36. RTCP Sender Report Format

  37. Inter-arrival Time Jitter

  38. Example for RTT computation

  39. Algorithm for RTCP Reporting Interval

  40. RTCP Bandwidth Allocation

  41. VoIP Phone Call Session Forward SIP signaling Forward RTP media Forward RTCP monitoring Reverse SIP signaling Callee Caller Reverse RTP media Reverse RTCP monitoring Logical channels between the caller and callee during a VoIP call

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