CSE 124 Networked Services Fall 2009

1. CSE 124 Networked ServicesFall 2009 B. S. Manoj, Ph.D http://cseweb.ucsd.edu/classes/fa09/cse124 Some of these slides are adapted from various sources/individuals including but not limited to the slides from the text books by Kurose and Ross. Use of these slides other than for pedagogical purpose for CSE 124, may require explicit permissions from the respective sources. CSE 124 Networked Services Fall 2009

2. Multimedia Networking Applications • Network applications can be broadly classified into • Loss sensitive • Data traffic such as HTTP or FTP traffic • Delay tolerant • Delay sensitive • Streamed stored audio/video • Streamed live audio/video • Interactive video • Loss tolerant • Loss and delay sensitive • Time-sensitive stock quotes • Health sensor traffic CSE 124 Networked Services Fall 2009

3. Streaming Stored Audio/Video • Streaming • The media transfer scheme where a part of the media file is played out while the remaining parts of the file are being received • Popular services: stored video sharing servers such s YouTube, Yahoo Videos, CNN etc. • Uni-directional media communication • Main features • Stored media files that are pre-recorded and coded • Streaming over the Internet • Streaming server pushes the content at a regular rate • Streaming client begins play back a few seconds after beginning reception • Two kinds of media players • Web browser-based and Host based • Continuous play out • Play out options are many: Fast Forward, Rewind, and Pause • Once play out begins, it should strive to maintain the original recorded timings • Key issue: getting the data over the network in time CSE 124 Networked Services Fall 2009

4. Streaming Live Audio and Video • Media source generates multimedia content in real-time • e.g., live video or audio transmission • Delay associated with content generation • Limited play out options: Limited Rewind and Pause • Uni-directional media communication • More stringent delay constraints than stored media streaming CSE 124 Networked Services Fall 2009

5. Real-time Interactive Audio/Video • Mostly bi-directional media communication • Each end-source generates media content in real-time • High delay constraints due to interactive nature of communication • End-to-end delay preferably < 150ms • e.g, Voice over IP applications such as Skype, Google Talk, Yahoo Messenger, Microsoft Netmeeting CSE 124 Networked Services Fall 2009

6. Why multimedia services are challenging? • Internet is designed for delay tolerant data communications • Best-effort traffic support only • Neither guarantee nor timeliness of data delivery • During high load situations • the delay performance can be worse • High load can be at the server, network links, or the routers • Main issues • Delay (latency or end-to-end delay) • Jitter (Delay jitter or Delay variation) • Packet loss CSE 124 Networked Services Fall 2009

7. Delay • Kinds of delays • Source delay (content generation delay • End-to-end delay • Play out delay • Source delay • Generating a media content takes certain amount of time 10101010000000…. 10101010000 Analog voice (4KHz) 8 KBytes per second 160 Bytes packet will take about 160 B/8KB/s= 20ms 120 B/8KB/s= 15ms Digitization (8KHz, 8 bits per sample) CSE 124 Networked Services Fall 2009

8. Delay (contd) • End-to-end delay • Due to the end-to-end network • Contributed by • Processing time by the intermediate routers • Queuing delay at intermediate routers • Transmission delay due to the source and intermediate routers • Propagation delay due to the links in the network CSE 124 Networked Services Fall 2009

9. 1. Router processing: Receive Check bit errors Buffer Determine output link transmission A propagation B Router processing queueing End-to-end delay: four sources • 2. Queueing • Time waiting at output link/buffer for transmission • Vary drastically depends on congestion level of router Introduction

10. 3. Transmission delay: R=link bandwidth (bps) L=packet length (bits) time to send bits into link = L/R 4. Propagation delay: d = length of physical link s = propagation speed in medium copper: ~2x108 m/sec Wireless: 3x 108 m/sec Fiber: 3x 108 m/sec propagation delay = d/s transmission A propagation B Router processing queueing End-to-end delay Introduction

11. R=link bandwidth (bps) L=packet length (bits) a=average packet arrival rate Queueing delay (revisited) traffic intensity = La/R • La/R ~ 0: average queueing delay small • La/R -> 1: delays become large • La/R > 1: more “work” arriving than can be serviced, average delay infinite! Introduction

12. End-to-end Delay • Delay at a router/node • End-to-end delay • dproc = processing delay • typically a few microsecs or less • dqueue = queuing delay • depends on congestion • dtrans = transmission delay • = L/R, significant for low-speed links • dprop = propagation delay • a few microsecs to hundreds of msecs • N = number of routers/nodes in the network • di= delay at router/node i CSE 124 Networked Services Fall 2009

13. Playout delay • The delay added/caused by the receiver-side media player • A certain amount of delay in playing out may improve the playout performance • Challenge is to get the required OS resources to play when desired • High priority for playout processes is essential • Two types • Fixed playout delay • Adaptive playout delay CSE 124 Networked Services Fall 2009

14. Jitter • The shared network resources such as links and routers • Results in high variability in end-to-end delay • Sometimes packets can be even out-of-ordered • Jitter cannot be easily removed • Because the network is best-effort • Its impact can be lessened • Receiver playout management 1 2 3 t t+20ms t+40ms 1 2 3 t+d t+20ms+2d t+40ms+d CSE 124 Networked Services Fall 2009

15. Handling Jitter • The impacts of Jitter can be managed together by • Sequence numbering • Time stamps • Receiver playout delay • Media source adds sequence numbers to every media packet • Sequence number increments with every packet • Usually unique for a certain duration of the session • Time stamps include the time instance at which the packets are generated • Sequence numbers and time stamps help • differentiate packet losses from silence periods CSE 124 Networked Services Fall 2009

16. 1 2 3 4 5 6 t t+20ms t+40ms t+80ms t+100ms t+120ms 1 2 3 4 5 6 Packet Loss t+d t+20ms+d t+40ms+d t+80ms+d t+100ms+d t+120ms+d 1 2 3 4 5 6 t t+20ms t+40ms t+60ms t+80ms t+100ms 1 2 3 6 t+d t+20ms+d t+40ms+d t+100ms+d Talk spurt 1 2 3 4 5 6 t t+20ms t+40ms t+60ms t+80ms t+100ms 1 2 3 4 5 6 CSE 124 Networked Services Fall 2009 t+d t+20ms+d t+40ms+d t+60ms+d t+80ms+d t+100ms+d

17. Receiver Playout delay • Delay added by receiver media player for every packet • Two approaches • Fixed playout delay • Adaptive playout delay • Fixed playout delay • Receiver fixes the playout delay for all packets • Simple to implement • e.g., media receiver plays out every packet exactly qunits of time after receiving it • If packet is received at time t, it is played at time t+q • Determining a good value for q is a challenge CSE 124 Networked Services Fall 2009

18. Fixed Playout Delay • sender generates packets every 20 msec during talk spurt. • first packet received at time r • first playout schedule: begins at p • second playout schedule: begins at p’ CSE 124 Networked Services Fall 2009

19. Determining Fixed Playout Delay • There are no strict rules for the choice of fixed playout delay • The delay is sufficient to handle the Jitter • One good estimate is the play out time can be equal to Mean Delay + Mean Jitter • Therefore, p = (Mean Delay + Mean Jitter) – r 0 ms 150 ms 400 ms CSE 124 Networked Services Fall 2009

20. Adaptive Playout Delay • In a dynamic network, Jitter can vary highly • Use of fixed playout delay can result in high packet loss or non optimal play out delay • Adaptive Playout delay is preferred in such dynamic situations • Adaptive playout delay, dynamically modifies the playout delay • Playout delay is modified based on the delay and jitter observations • Playout delay is estimated for every packet, however, modified only when the talk spurt begins • Objective: Minimize playout delay, keeping late loss rate low CSE 124 Networked Services Fall 2009

21. Estimating Adaptive Playout Delay • One Approach to adaptive playout delay adjustment: • estimate network delay, adjust playout delay at beginning of each talk spurt. • silent periods compressed and elongated. • chunks still played out every 20 msec during talk spurt. dynamic estimate of average delay at receiver: where u is a fixed constant (e.g., u = .01). CSE 124 Networked Services Fall 2009

22. Estimating Adaptive playout delay • also useful to estimate average deviation of delay, vi : • estimates di , vi calculated for every received packet (but used only at start of talk spurt • for first packet in talk spurt, playout time is: • where K is positive constant • remaining packets in talkspurt are played out periodically at time CSE 124 Networked Services Fall 2009

23. Transport layer choice for multi-media applications • TCP-like transport protocols are not suitable for multimedia traffic • They are connection oriented • high overhead • They offer reliable delivery • high delay due to potential retransmissions • Larger playout delay: smooth TCP delivery rate • HTTP/TCP passes more easily through firewalls • Transmission rate fluctuates due to TCP congestion control • UDP-like light connection less protocols are preferred • Low end-to-end delay • short playout delay (2-5 seconds) to remove network jitter • Due to administrative reasons, TCP still dominates the multimedia video/audio transport • UDP is prominent for VoIP applications CSE 124 Networked Services Fall 2009

24. Packet loss • Packet loss is unavoidable • Recovery from packet loss is an important objective • Lossy recovery is sufficient for multimedia • Two popular approaches • Forward Error Correction • Packet Interleaving CSE 124 Networked Services Fall 2009

25. Approach: Packet Redundancy for every group of N chunks create redundant chunk by exclusive OR-ing N original chunks send out N+1 chunks, increasing bandwidth by factor 1/N. can reconstruct original N chunks if at most one lost chunk from N+1 chunks playout delay: enough time to receive all N+1 packets tradeoff: increase N, less bandwidth waste increase N, longer playout delay increase N, higher probability that 2 or more chunks will be lost Forward Error Correction

26. FEC • Approach: Stream redundancy • “piggyback lower quality stream” • send lower resolutionaudio stream as redundant information • e.g., nominal stream PCM at 64 kbpsand redundant streamGSM at 13 kbps. • whenever there is non-consecutive loss, receiver can conceal the loss. • can also append (n-1)st and (n-2)nd low-bit ratechunk

27. Interleaving chunks divided into smaller units for example, four 5 msec units per chunk packet contains small units from different chunks if packet lost, still have most of every chunk no redundancy overhead, but increases playout delay Packet Interleaving method

28. Week-2-Homework • Reading assignments • File Transfer protocol • End-of-chapter Problems P10 and P11 from Chapter 7 of Kurose and Ross (page 676) • Will be placed at the course website CSE 124 Networked Services Fall 2009