Mul$media Networking. #3 Mul$media Networking Semester Ganjil PTIIK Universitas Brawijaya. #3 Requirements of Mul$media Networking

Transcription

1 Mul$media #3 Mul$media Semester Ganjil PTIIK Universitas Brawijaya

2 Schedule of Class Mee$ng 1. Introduc$on 2. Applica$ons of MN 3. Requirements of MN 4. Coding and Compression 5. RTP 6. IP Mul$cast 7. IP Mul$cast (cont d) 8. Overlay Mul$cast 9. CDN: Solu$ons 10. CDN: Case Studies 11. QoS on the Internet: Constraints 12. QoS on the Internet: Solu$ons 13. Discussion 14. Summary

3 Today s Outline Requirements of Mul$media Client- side buffering Removing Ji[er Recovering from Packet Loss Coding and Compression Digital Audio Digital Video

4 Streaming stored video: Cumulative data 1. video recorded (e.g., 30 frames/sec) 2. video sent network delay (fixed in this example) 3. video received, played out at client (30 frames/sec) time streaming: at this time, client playing out early part of video, while server still sending later part of video

5 Streaming stored video: systems UDP Streaming unpredictable available bandwidth constant- rate streaming can fail requires a media control server interac$vity many firewalls block UDP traffic HTTP Streaming & Adap$ve Streaming TCP conges$on control & retransmission delay playout client- side buffering & prefetching smooth playout Majority of today s systems employ HTTP streaming and adap$ve HTTP streaming

6 Streaming stored video: revisited constant bit rate video transmission Cumulative data t 0 t 0 +2Δ t 0 +Δ Δ = fixed $me, each video block played out $me time

7 Streaming stored video: revisited constant bit rate video transmission client video reception Cumulative data variable network delay t 0 t 0 +2Δ t 1 t 0 +Δ t 2 t 1 +Δ variable end- to- end delays, each video block may experience different delays time

8 Streaming stored video: revisited Cumulative data t 0 constant bit rate video transmission variable network delay t 0 +2Δ t 1 t 0 +Δ client video reception client playout delay t 2 t 3 t 1 +Δ buffered video t 3 +Δ Client- side buffering is used to mi$gate: varying end- to- end delays varying available bandwidth constant bit rate video playout at client time

9 Today s Outline Requirements of Mul$media Client- side buffering Removing Ji;er Recovering from Packet Loss Coding and Compression Digital Audio Digital Video

10 Removing Ji[er in VoIP Ji[er can be removed using: prepend each chunk with sequence number & +mestamp, delaying playout VoIP characteris$cs: 64 kbps during talk spurt pkts generated only during talk spurts 20 msec chunks at 8 Kbytes/sec: 160 bytes of data typical maximum tolerable delay: 400 ms

11 Delay ji[er Cumulative data constant bit rate transmission variable network delay (jitter) client reception buffered data constant bit rate playout at client client playout delay end- to- end delays of two consecu$ve packets: difference can be more or less than 20 msec (transmission $me difference) time

12 VoIP: fixed playout delay receiver a[empts to playout each chunk exactly q msecs ajer chunk was generated. chunk has $me stamp t: play out chunk at t+q chunk arrives ajer t+q: data arrives too late for playout: data lost tradeoff in choosing q: large q: less packet loss small q: be[er interac$ve experience

13 VoIP: fixed playout delay sender generates packets every 20 msec during talk spurt. first packet received at $me r first playout schedule: begins at p second playout schedule: begins at p packets Two policies, wait p or wait p - p has less delay, but one missed - p has no missed, but higher delay packets generated packets received loss playout schedule p' - r playout schedule p - r time r p p'

14 Adap$ve playout delay (1) goal: low playout delay, low late loss rate approach: adap$ve playout delay adjustment: es$mate network delay, adjust playout delay at beginning of each talk spurt silent periods compressed and elongated chunks s$ll played out every 20 msec during talk spurt adap$vely es$mate packet delay: (EWMA - exponen$ally weighted moving average, recall TCP RTT es$mate): d i = (1-α)d i-1 + α (r i t i ) delay estimate after ith packet small constant, e.g. 0.1 of ith packet time received - time sent (timestamp) measured end-to-end delay

15 Adap$ve playout delay (2) also useful to es$mate average devia$on of delay, v i : v i = (1-β)v i-1 + β r i t i d i es$mates d i, v i calculated for every received packet, but used only at start of talk spurt for first packet in talk spurt, playout $me is: playout-time i = t i + d i + Kv i remaining packets in talkspurt are played out periodically

16 Adap$ve playout delay (3) Q: How does receiver determine whether packet is first in a talk spurt? if no loss, receiver looks at successive $mestamps difference of successive stamps > 20 msec talk spurt begins. with loss possible, receiver must look at both $me stamps and sequence numbers difference of successive stamps > 20 msec and sequence numbers without gaps - - > talk spurt begins.

17 Today s Outline Requirements of Mul$media Client- side buffering Removing Ji[er Recovering from Packet Loss Coding and Compression Digital Audio Digital Video

18 VoiP: recovery from packet loss (1) Challenge: recover from packet loss given small tolerable delay between original transmission and playout each ACK/NAK takes ~ one RTT alterna$ve: Forward Error Correc+on (FEC) send enough bits to allow recovery without retransmission simple FEC for every group of n chunks, create redundant chunk by exclusive OR- ing n original chunks send n+1 chunks, increasing bandwidth by factor 1/n can reconstruct original n chunks if at most one lost chunk from n +1 chunks, with playout delay

19 VoIP: Loss Encode Transmit Decode

20 VoIP: Loss Encode 1 4 Transmit Decode

21 VoIP: Loss Encode 1 4 Transmit 1?????? 4 Decode What to do about the missing packets?

22 VoIP: Recovering from Loss Encode Transmit Decode

23 VoIP: Recovering from Loss Encode Transmit Decode

24 VoiP: recovery from packet loss (2) another FEC scheme: piggyback lower quality stream send lower resolu$on audio stream as redundant informa$on e.g., nominal stream PCM at 64 kbps and redundant stream GSM at 13 kbps non- consecu$ve loss: receiver can conceal loss generaliza$on: can also append (n- 1)st and (n- 2)nd low- bit rate chunk

25 VoiP: recovery from packet loss (3) interleaving to conceal loss: audio chunks divided into smaller units, e.g. four 5 msec units per 20 msec audio chunk packet contains small units from different chunks if packet lost, s$ll have most of every original chunk no redundancy overhead, but increases playout delay

26 End of Today s Lecture Do you have any ques$on?