1. CSE 124 Networked Services Fall 2009 B. S. Manoj, Ph.D http://cseweb.ucsd.edu/classes/fa09/cse124 10/8/20091CSE 124 Networked Services Fall 2009 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.

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 10/8/20092CSE 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 10/8/20093CSE 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 10/8/20094CSE 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 10/8/20095CSE 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 10/8/20096CSE 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 10/8/20097CSE 124 Networked Services Fall 2009 Analog voice (4KHz) Digitization (8KHz, 8 bits per sample) 10101010000000…. 8 KBytes per second 10101010000 160 Bytes packet will take about 160 B/8KB/s= 20ms 120 B/8KB/s= 15ms

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 10/8/20098CSE 124 Networked Services Fall 2009

9. Introduction1-9 End-to-end delay: four sources 1. Router processing: – Receive – Check bit errors – Buffer – Determine output link A B propagation transmission Router processing queueing  2. Queueing  Time waiting at output link/buffer for transmission  Vary drastically depends on congestion level of router

10. Introduction1-10 End-to-end delay 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: ~2x10 8 m/sec – Wireless: 3x 10 8 m/sec – Fiber: 3x 10 8 m/sec propagation delay = d/s A B propagation transmission Router processing queueing

11. Introduction1-11 Queueing delay (revisited) R=link bandwidth (bps) L=packet length (bits) a=average packet arrival rate 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!

12. End-to-end Delay Delay at a router/node End-to-end delay 10/8/2009CSE 124 Networked Services Fall 200912 d proc = processing delay – typically a few microsecs or less d queue = queuing delay – depends on congestion d trans = transmission delay – = L/R, significant for low- speed links d prop = propagation delay – a few microsecs to hundreds of msecs – N = number of routers/nodes in the network – di = delay at router/node i

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 10/8/200913CSE 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 t 12 1 t+d t+20ms t+20ms+2d 2 3 t+40ms 3 t+40ms+d 10/8/200914CSE 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 10/8/200915CSE 124 Networked Services Fall 2009

16. t 12 1 t+d t+20ms t+20ms+d 2 3 t+40ms 3 456 456 t+80mst+100mst+120ms t+40ms+dt+80ms+dt+100ms+dt+120ms+d t 12 1 t+d t+20ms t+20ms+d 2 3 t+40ms 3 456 6 t+60mst+80mst+100ms t+40ms+dt+100ms+d t 12 1 t+d t+20ms t+20ms+d 2 3 t+40ms 3 456 456 t+60mst+80mst+100ms t+40ms+dt+60ms+dt+80ms+dt+100ms+d Packet Loss Talk spurt 10/8/200916CSE 124 Networked Services Fall 2009

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 q units 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 10/8/200917CSE 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’ 10/8/2009

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 150 ms400 ms 0 ms 10/8/200919CSE 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 10/8/200920CSE 124 Networked Services Fall 2009

21. Estimating Adaptive Playout Delay dynamic estimate of average delay at receiver: where u is a fixed constant (e.g., u =.01). 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. 10/8/200921CSE 124 Networked Services Fall 2009

22. Estimating Adaptive playout delay  also useful to estimate average deviation of delay, v i :  estimates d i, v i 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 10/8/200922CSE 124 Networked Services Fall 2009

23. 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 Transport layer choice for multi- media applications 10/8/200923CSE 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 10/8/2009CSE 124 Networked Services Fall 200924

25. Forward Error Correction 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

26. FEC Approach: Stream redundancy  “piggyback lower quality stream”  send lower resolution audio stream as redundant information  e.g., nominal stream PCM at 64 kbps and redundant stream GSM 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 rate chunk

27. Packet Interleaving method 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

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 10/8/200928CSE 124 Networked Services Fall 2009