1. TCP/IP Protocol Suite 1 Chapter 25 Upon completion you will be able to: Multimedia Know the characteristics of the 3 types of services Understand the methods of digitizing and compressing. Understand jitter, translation, and mixing in real-time traffic Understand the role of RTP and RTCP in real-time traffic Understand how the Internet can be used as a telephone network Objectives

2. TCP/IP Protocol Suite 2 Figure 25.1 Internet audio/video

3. TCP/IP Protocol Suite 3 Streaming stored audio/video refers to on-demand requests for compressed audio/video files. Note:

4. TCP/IP Protocol Suite 4 Streaming live audio/video refers to the broadcasting of radio and TV programs through the Internet. Note:

5. TCP/IP Protocol Suite 5 Interactive audio/video refers to the use of the Internet for interactive audio/video applications. Note:

6. TCP/IP Protocol Suite 6 What is Sound Sound is a wave that involves a molecules of air being compressed and expanded under the action of a physical device. Without air there is no sound Sound is a pressure wave that has continuous values Sound is measured by measuring the wave pressure at any point. Trnsducer (a device that transforms one type of energy into another) converts pressure to voltage level Voltages’ amplitude change over time (pressure increases or decreases)

7. TCP/IP Protocol Suite 7 Digitization The voltage levels (amplitude) are represented by a continues signal (analog) that varies over time (two dimensions: amplitude and time) To digitize an analog signal (Convert it into a digital signal) we sample in each dimension The rate at which we sample is referred to as sampling frequency

8. TCP/IP Protocol Suite 8 Digitization Audio sampling rate (8 kHz to 48 kHz) Human ear hear between 20 Hz to 20 kHz Ultrasound > 20 kHz Human voice can reach approximately 4 kHz Sampling in the amplitude dimension is called quantization Typical uniform quantization rate is 8-bit and 16-bit 8-bit quantization divides the vertical axis into 256 (2 8 ) levels, and 16-bit divides it into (2 16 ) 65,536 levels Voice is sampled at 8000 samples per second (8 kHz) with 8 bits per sample, this results in a digital signal of 64 kbps. Music is sampled at 44,100 samples per second (44.1 kHz) with 16 bits per sample, this results in a digital signal of 705.6 kbps for monaural and 1.411 Mbps for stereo.

9. TCP/IP Protocol Suite 9 25.1 DIGITIZING AUDIO AND VIDEO Before audio or video signals can be sent on the Internet, they need to be digitized. We discuss audio and video separately. The topics discussed in this section include: Digitizing Audio Digitizing Video

10. TCP/IP Protocol Suite 10 Compression is needed to send video over the Internet. Note:

11. TCP/IP Protocol Suite 11 25.2 AUDIO AND VIDEO COMPRESSION To send audio or video over the Internet requires compression. The topics discussed in this section include: Audio Compression Video Compression

12. TCP/IP Protocol Suite 12 Figure 25.2 JPEG gray scale

13. TCP/IP Protocol Suite 13 Figure 25.3 JPEG process

14. TCP/IP Protocol Suite 14 Figure 25.4 Case 1: uniform gray scale

15. TCP/IP Protocol Suite 15 Figure 25.5 Case 2: two sections

16. TCP/IP Protocol Suite 16 Figure 25.6 Case 3: gradient gray scale

17. TCP/IP Protocol Suite 17 Figure 25.7 Reading the table

18. TCP/IP Protocol Suite 18 Figure 25.8 MPEG frames

19. TCP/IP Protocol Suite 19 Figure 25.9 MPEG frame construction

20. TCP/IP Protocol Suite 20 25.3 STREAMING STORED AUDIO/VIDEO We turn our attention to a specific applications called streaming stored audio and video. We use four approaches to show how a file can be downloaded, each with a different complexity. The topics discussed in this section include: First Approach: Using a Web Server Second Approach: Using a Web Server with Metafile Third Approach: Using a Media Server Fourth Approach: Using a Media Server and RTSP

21. TCP/IP Protocol Suite 21 Figure 25.10 Using a Web server

22. TCP/IP Protocol Suite 22 Figure 25.11 Using a Web server with a metafile

23. TCP/IP Protocol Suite 23 Figure 25.12 Using a media server

24. TCP/IP Protocol Suite 24 Figure 25.13 Using a media server and RTSP

25. TCP/IP Protocol Suite 25 25.4 STREAMING LIVE AUDIO/VIDEO In streaming live audio/video the stations broadcast through the Internet. Communication is multicast and live. Live streaming is better suited to the multicast services of IP and the use of protocols such as UDP and RTP.

26. TCP/IP Protocol Suite 26 25.5 REAL-TIME INTERACTIVE AUDIO/VIDEO In real-time interactive audio/video, people communicate visually and orally with one another in real time. Examples include video conferencing and the Internet phone or voice over IP. The topics discussed in this section include: Characteristics

27. TCP/IP Protocol Suite 27 Figure 25.14 Time relationship

28. TCP/IP Protocol Suite 28 Jitter is introduced in real-time data by the delay between packets. Note:

29. TCP/IP Protocol Suite 29 Figure 25.15 Jitter

30. TCP/IP Protocol Suite 30 Figure 25.16 Timestamp

31. TCP/IP Protocol Suite 31 To prevent jitter, we can timestamp the packets and separate the arrival time from the playback time. Note:

32. TCP/IP Protocol Suite 32 Figure 25.17 Playback buffer

33. TCP/IP Protocol Suite 33 A playback buffer is required for real-time traffic. Note:

34. TCP/IP Protocol Suite 34 A sequence number on each packet is required for real-time traffic. Note:

35. TCP/IP Protocol Suite 35 Real-time traffic needs the support of multicasting. Note:

36. TCP/IP Protocol Suite 36 Translation means changing the encoding of a payload to a lower quality to match the bandwidth of the receiving network. Note:

37. TCP/IP Protocol Suite 37 Mixing means combining several streams of traffic into one stream. Note:

38. TCP/IP Protocol Suite 38 TCP, with all its sophistication, is not suitable for interactive multimedia traffic because we cannot allow retransmission of packets. Note:

39. TCP/IP Protocol Suite 39 UDP is more suitable than TCP for interactive traffic. However, we need the services of RTP, another transport layer protocol, to make up for the deficiencies of UDP. Note:

40. TCP/IP Protocol Suite 40 25.6 RTP Real-time Transport Protocol (RTP) is the protocol designed to handle real-time traffic on the Internet. RTP does not have a delivery mechanism; it must be used with UDP. The topics discussed in this section include: RTP Packet Format UDP Port

41. TCP/IP Protocol Suite 41 Figure 25.18 RTP

42. TCP/IP Protocol Suite 42 Figure 25.19 RTP packet header format

43. TCP/IP Protocol Suite 43 Table 25.1 Payload types

44. TCP/IP Protocol Suite 44 RTP uses a temporary even-numbered UDP port. Note:

45. TCP/IP Protocol Suite 45 25.7 RTCP Real-time Transport Control Protocol (RTCP) is a protocol that allows messages that control the flow and quality of data. RTCP has five types of messages. The topics discussed in this section include: Sender Report Receiver Report Source Description Message Bye Message Application Specific Message UDP Port

46. TCP/IP Protocol Suite 46 Figure 25.20 RTCP message types

47. TCP/IP Protocol Suite 47 RTCP uses an odd-numbered UDP port number that follows the port number selected for RTP. Note:

48. TCP/IP Protocol Suite 48 25.8 VOICE OVER IP Voice over IP, or Internet telephony is an application that allows communication between two parties over the packet-switched Internet. Two protocols have been designed to handle this type of communication: SIP and H.323. The topics discussed in this section include: SIPH.323

49. TCP/IP Protocol Suite 49 Figure 25.21 SIP messages

50. TCP/IP Protocol Suite 50 Figure 25.22 SIP formats

51. TCP/IP Protocol Suite 51 Figure 25.23 SIP simple session

52. TCP/IP Protocol Suite 52 Figure 25.24 Tracking the callee

53. TCP/IP Protocol Suite 53 Figure 25.25 H.323 architecture

54. TCP/IP Protocol Suite 54 Figure 25.26 H.323 protocols

55. TCP/IP Protocol Suite 55 Figure 25.27 H.323 example