Date post: | 21-Dec-2015 |
Category: |
Documents |
View: | 213 times |
Download: | 0 times |
TCP/IP Protocol Suite 1
Chapter 25Chapter 25
Upon completion you will be able to:
MultimediaMultimedia
• 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
TCP/IP Protocol Suite 2
Figure 25.1 Internet audio/video
TCP/IP Protocol Suite 3
Streaming stored audio/video refers to on-demand requests for compressed
audio/video files.
Note:Note:
TCP/IP Protocol Suite 4
Streaming live audio/video refers to the broadcasting of radio and TV programs through the Internet.
Note:Note:
TCP/IP Protocol Suite 5
Interactive audio/video refers to the use of the Internet for interactive
audio/video applications.
Note:Note:
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)
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
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 (28)
levels, and 16-bit divides it into (216) 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.
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 Before audio or video signals can be sent on the Internet, they need to be digitized. We discuss audio and video separately.digitized. We discuss audio and video separately.
The topics discussed in this section include:The topics discussed in this section include:
Digitizing Audio Digitizing Audio Digitizing Video Digitizing Video
TCP/IP Protocol Suite 10
Compression is needed to send video over the Internet.
Note:Note:
TCP/IP Protocol Suite 11
25.2 AUDIO AND VIDEO COMPRESSION
To send audio or video over the Internet requires compression. To send audio or video over the Internet requires compression.
The topics discussed in this section include:The topics discussed in this section include:
Audio Compression Audio Compression Video Compression Video Compression
TCP/IP Protocol Suite 12
Figure 25.2 JPEG gray scale
TCP/IP Protocol Suite 13
Figure 25.3 JPEG process
TCP/IP Protocol Suite 14
Figure 25.4 Case 1: uniform gray scale
TCP/IP Protocol Suite 15
Figure 25.5 Case 2: two sections
TCP/IP Protocol Suite 16
Figure 25.6 Case 3: gradient gray scale
TCP/IP Protocol Suite 17
Figure 25.7 Reading the table
TCP/IP Protocol Suite 18
Figure 25.8 MPEG frames
TCP/IP Protocol Suite 19
Figure 25.9 MPEG frame construction
TCP/IP Protocol Suite 20
25.3 STREAMING STORED AUDIO/VIDEO
We turn our attention to a specific applications called streaming stored 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 audio and video. We use four approaches to show how a file can be downloaded, each with a different complexity.downloaded, each with a different complexity.
The topics discussed in this section include:The topics discussed in this section include:
First Approach: Using a Web Server First Approach: Using a Web Server Second Approach: Using a Web Server with Metafile Second Approach: Using a Web Server with Metafile Third Approach: Using a Media Server Third Approach: Using a Media Server Fourth Approach: Using a Media Server and RTSP Fourth Approach: Using a Media Server and RTSP
TCP/IP Protocol Suite 21
Figure 25.10 Using a Web server
TCP/IP Protocol Suite 22
Figure 25.11 Using a Web server with a metafile
TCP/IP Protocol Suite 23
Figure 25.12 Using a media server
TCP/IP Protocol Suite 24
Figure 25.13 Using a media server and RTSP
TCP/IP Protocol Suite 25
25.4 STREAMING LIVE AUDIO/VIDEO
In streaming live audio/video the stations broadcast through the In streaming live audio/video the stations broadcast through the Internet. Communication is multicast and live. Live streaming is better Internet. Communication is multicast and live. Live streaming is better suited to the multicast services of IP and the use of protocols such as suited to the multicast services of IP and the use of protocols such as UDP and RTP.UDP and RTP.
TCP/IP Protocol Suite 26
25.5 REAL-TIME INTERACTIVE AUDIO/VIDEO
In real-time interactive audio/video, people communicate visually and In real-time interactive audio/video, people communicate visually and orally with one another in real time. Examples include video orally with one another in real time. Examples include video conferencing and the Internet phone or voice over IP. conferencing and the Internet phone or voice over IP.
The topics discussed in this section include:The topics discussed in this section include:
Characteristics Characteristics
TCP/IP Protocol Suite 27
Figure 25.14 Time relationship
TCP/IP Protocol Suite 28
Jitter is introduced in real-time data by the delay between packets.
Note:Note:
TCP/IP Protocol Suite 29
Figure 25.15 Jitter
TCP/IP Protocol Suite 30
Figure 25.16 Timestamp
TCP/IP Protocol Suite 31
To prevent jitter, we can timestamp the packets and separate the arrival time
from the playback time.
Note:Note:
TCP/IP Protocol Suite 32
Figure 25.17 Playback buffer
TCP/IP Protocol Suite 33
A playback buffer is required forreal-time traffic.
Note:Note:
TCP/IP Protocol Suite 34
A sequence number on each packet is required for real-time traffic.
Note:Note:
TCP/IP Protocol Suite 35
Real-time traffic needs the support of multicasting.
Note:Note:
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:Note:
TCP/IP Protocol Suite 37
Mixing means combining several streams of traffic into one stream.
Note:Note:
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:Note:
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:Note:
TCP/IP Protocol Suite 40
25.6 RTP
Real-time Transport Protocol (RTP) is the protocol designed to handle Real-time Transport Protocol (RTP) is the protocol designed to handle real-time traffic on the Internet. RTP does not have a delivery real-time traffic on the Internet. RTP does not have a delivery mechanism; it must be used with UDP.mechanism; it must be used with UDP.
The topics discussed in this section include:The topics discussed in this section include:
RTP Packet Format RTP Packet Format UDP Port UDP Port
TCP/IP Protocol Suite 41
Figure 25.18 RTP
TCP/IP Protocol Suite 42
Figure 25.19 RTP packet header format
TCP/IP Protocol Suite 43
Table 25.1 Table 25.1 Payload typesPayload types
TCP/IP Protocol Suite 44
RTP uses a temporary even-numbered UDP port.
Note:Note:
TCP/IP Protocol Suite 45
25.7 RTCP
Real-time Transport Control Protocol (RTCP) is a protocol that allows Real-time Transport Control Protocol (RTCP) is a protocol that allows messages that control the flow and quality of data. RTCP has five types messages that control the flow and quality of data. RTCP has five types of messages.of messages.
The topics discussed in this section include:The topics discussed in this section include:
Sender Report Sender Report Receiver Report Receiver Report Source Description Message Source Description Message Bye Message Bye Message Application Specific Message Application Specific Message UDP Port UDP Port
TCP/IP Protocol Suite 46
Figure 25.20 RTCP message types
TCP/IP Protocol Suite 47
RTCP uses an odd-numbered UDP port number that follows the port
number selected for RTP.
Note:Note:
TCP/IP Protocol Suite 48
25.8 VOICE OVER IP
Voice over IP, or Internet telephony is an application that allows Voice over IP, or Internet telephony is an application that allows communication between two parties over the packet-switched Internet. communication between two parties over the packet-switched Internet. Two protocols have been designed to handle this type of communication: Two protocols have been designed to handle this type of communication: SIP and H.323. SIP and H.323.
The topics discussed in this section include:The topics discussed in this section include:
SIP SIP H.323 H.323
TCP/IP Protocol Suite 49
Figure 25.21 SIP messages
TCP/IP Protocol Suite 50
Figure 25.22 SIP formats
TCP/IP Protocol Suite 51
Figure 25.23 SIP simple session
TCP/IP Protocol Suite 52
Figure 25.24 Tracking the callee
TCP/IP Protocol Suite 53
Figure 25.25 H.323 architecture
TCP/IP Protocol Suite 54
Figure 25.26 H.323 protocols
TCP/IP Protocol Suite 55
Figure 25.27 H.323 example