1. Ch 6. Multimedia Networking Myungchul Kim mckim@icu.ac.kr

2. 2 o Networked multimedia applications: timing and tolerance of data loss o Delay-sensitive and loss-tolerant o Streaming stored audio and video – Stored media – Streaming: avoids having to download the entire file before beginning playout. Realplayer, QuickTime and Media Player – Continuous playout o Streaming live audio and video – Not stored, not fast-forward – Use the IP multicast

3. 3 o Real-time interactive audio and video – Real-time – Interactive – Internet phone – For voice, 150 msec, 150-400 msec, 400 msec o Hurdles for multimedia – End-to-end delay for a packet – Variation of packet delay, packet jitter – Packet loss o Supporting multimedia better in Internet – Reservation approach – Laissez-faire approach: ISP, CDN, multicast overlay networks – Differentiation approach

4. 4 o Audio and video compression – 1024 pixels * 1024 pixels with each pixel encoded into 24 bits => 3 Mbyte – 7 Min over a 64 kbps link – If the image is compressed at 10:1, o Audio compression – 8000 samples per second -> quantization with 256 values (8bits)-> 64,000 bits/second – Pulse code modulation – GSM, G.729, MPEG 1 layer 3(MP3),… o Video compression – MPEG 1, 2, 4, H.261

5. 5 o Real-time streaming protocol (RTSP) – User interactivity – RealPlayer and Media Player – Decompression, jitter removal, and correction – Fig 6.2 Streaming stored audio and video

6. 6 o Fig 6.3

7. 7 o RTSP – Control the playback of continuous media – No related with compression schemes, encapsulation in packets, transportation, buffering – Out-of-band protocol – Over either TCP or UDP – Pause/resume, playback, fast-forward, and rewind

8. 8 o Fig 6.5

9. 9 o Every 20 msec over UDP o Packet loss, end-to-end delay, and packet jitter o Removing jitter at the receiver for audio – With a sequence number, a timestamp or – delaying playout at the receiver Internet phone

10. 10 o Fig 6.6

11. 11 o RTP – For sound and video – On UDP – RTP header: the type of audio encoding, a sequence number, and a timestamp – Sequence number: detect packet loss – Timestamp: synchronous playout at the receiver – Synchronization source identifier (SSRC): identify the source of the RTP stream – Fig 6.9 Protocols for real-time interactive applications

12. 12 o Table 6.1 and 6.2

13. 13 o Developing software applications with RTP o Fig 6.10 and 6.11

14. 14 o RTP control protocol (RTCP) – In conjunction with RTP – Report statistics including number of packets sent, number of packets lost, and interarrival jitter. – RTCP (5 % of the session bandwidth). – Fig 6.12

15. 15 o SIP – Establish calls between a caller and a callee over an IP network – Caller determines the current IP address of the callee – Call management (encoding, new participants, call transfer,…) – Fig 6.13

16. 16 o SIP – SIP proxy and registrar (cf. DNS) – Fig 6.14

17. 17 o H.323 – Fig 6.15