Multimedia lecture6

Eng: Mohammed Hussein1
Republic of Yemen
THAMAR UNIVERSITY
Faculty of Computer Science&
Information System
Lecturer, and Researcher atThamar University
By Eng: Mohammed Hussein

Outline
 Introduction
 Types of multimedia services
1. Streaming stored audio/video
 Video on demand
2. Streaming Live audio/video
 Direct to home (DTH)
3. Interactive real time audio/video
 Teleconferencing
 Voice over IP
2 Eng: Mohammed Hussein

Introduction of multimedia services
The deployment of high-speed networks and reduction in bandwidth
requirement has led to the emergence diverse of many applications.
Because the audio/video compression.
Classes of audio/video services applications:
1) Streaming stored audio and video
2) Streaming live audio and video
3) Real-time interactive audio and video
Fundamental characteristics:
 Typically delay sensitive
 end-to-end delay
 delay jitter
 But loss tolerant

Streaming stored audio/video
 Approach 1: compressed audio/video file is downloaded just like a
text file by a client.The client then plays the file.
1. The uncompressed file become larger,
example: 1-hour the size approximately
about 600Mb and the file after
compression 300 Mb forVCR quality 1-
hour video using MPEG1.Then it can be
played.
2. The downloading will depending on the
size of file and bandwidth.
3. It will reserve at client side larger
storage space.
4. So, it doesn’t provide a streaming stored
services.
 Limitation:
1: GET: audio/video file
2: RESPONSE
Media
player
3: audio/video file
Web
server
Server
Browser
Client

 Approach 2: the media player is directly connected to the web server for
downloading the audio/video file.The media player uses the metafile to
interact direct with web server and the web server response to the client
via media player.
The advantage of this approach:
1. The storage requirement at the
client side is small.
2. We are using two different type
of files: the metafile and the
audio/video file.
3. The client can play without
delay of downloading.
• Limitation here is both the browser and media player uses HTTP protocol for downloading
the two files. So HTTP runs overTCP and all data will transformed usingTCP. TheTCP is
not good protocol to provide streaming audio/video services. (flow control, error control,.)
1: GET: metafile
2: RESPONSE
Media
player
3: metafile
Web
server
Server
Browser
Client
5: RESPONSE

 Approach 3: A separate media server is used for downloading the
audio/video file.
 Advantage: avoids the use of TCP, which is unsuitable for
downloading audio/video files.
 The browser communicate with
web server and send metafile to
media player.
 The media player use the metafile
to communicate with a media
server to get the audio/video file.
 This approach uses UDP and RTP
protocols as we discussed in
lecture5.
1: GET: metafile
2: RESPONSE
Media
player
3: metafile
Web
Server
Server
Browser
Client
5: RESPONSE
Media
Server

Video on demand application
 As the user can play, pause and replay .The storage video in the CD or DVD.
 So, we would like to do that in the network as the requirements of VOD
7
Eng: Mohammed Hussein
Video
Server
Server
Audio
Server
Server
Switch
High Speed Network
(SONET,ATM,
MPLS,LTE, …)
Local distribution Network
(LAN,WLAN …)
Switch

VOD Requirements of educational
 The video-on-demand service at the campus.
 Infrastructure deployed:
1. High speed LAN (Gigabit Ethernet) and ADSL
2. Media Servers
3. Software:
 OS- windows 2008/.NET server
 Encoding software- windows media encoding at the server.
 Windows media player at the server.
 The above two are free and provides good quality audio/video above
128 Kbps.

The Distribution Network
 This example of one university
 Gigabit Ethernet based backbone network in the institutional area
 DSL based broadband access in the residential area.
 The DSLAM at the access provider is the equipment that really allows DSL
to happen.A DSLAM takes connections from many customers and
aggregates them onto a single, high-capacity connection to the Internet.

Media servers
10
 Media servers has to be powerful in terms of Processor and Hard
disk.An example
 Processor: P-IV(1.3 GHz), Main memory: 1 GB
 Hard disk: RAID-5 (4:1, 147 GB each)

What is Flash Media Server (FMS)?
 Adobe Flash Media Server is a real-time media server
 It can deliver live audio/video, stream audio/video, record
audio/video etc .
 Flash Media Server Applications
 Live video broadcasting
 Interactive gaming
 Video conferencing

How the FMS works?
1. Client-sideAction Script file (.swf)
resides in theWeb Server.
2. Server-sideAction Script file (main.asc)
may or may not reside in the same
machine
3. Client makes connection to Flash Media
Server via RealTime Messaging
Protocol (RTMP).
4. Web Server sends swf file to flash client
over HTTP.
5. Client plays the swf file
6. Client connects to FMS using RTMP
7. FMS and client communicates via
RTMP.
Client-server architecture12 Eng: Mohammed Hussein

Storage issue related to VOD services
 When we speak about 100 courses each one 1-hour, so storing all
courses in a Hard disk using RAD5.
 RAD5 which provides higher throughput that is necessary for
streaming purposes and we can Use 3 media server with the same IP.
 Because all courses are not equally popular. One possible solution is to
use memory hierarchy.
 Another approach is to distribute the courses in multiple servers with
the same IP.
costcapacity

Real time issue related to VOD services
Example the time of encoding here request rate of 766 Kbps to get
flicker free display.
 So the output streams to meet timing requirements (766 Kbps).
As we know to read data from disks it is read in terms of sectors.
In other hand when it is display it is done in streaming way ( continues
manner).Therefore, we should use buffering : (Read one sector and One
Sector transmitting)
Read one sector
One Sector transmitting
The data here is sent continuously
Data read from disk
Transmission
Buffered data

Streaming Live Audio/Video
 SatelliteTelevision distribution system (or Broadcasting of radio/TV
program)
 The user get the services through dish with a broadcast of frequency
 DirectTo Home (DTH) services
 The user use small antenna which provided by the satellite company to
get the services directly with a help of set top box (receiver).
 Set top box(receiver) is like a computer which has CPU, RAM to
MPEG, NIC to network, I/O toTV and remote control

Real-Time Interactive Audio/Video
 In real-time interactive audio/video, people communicate with one
another in real time, an example that allows people to communicate
visually and orally, the applications are:
1. The Internet phone or voice over IP
2. Video conferencing
 Characteristics:
 Before addressing the protocols used in this class of applications, we
discuss some characteristics of real-time audio/video communication.
1. Time Relationship
2. Timestamp
3. Playback Buffer
4. Ordering

Characteristic : Time Relationship
17
 The time relationship between the packets is preserved.
 Real-time data on a packet-switched network require the preservation of the
time relationship between packets of a session. For example, let us assume
that a real-time video server creates live video images and sends them
online.The video is digitized and packetized.
 There are only three packets, and each packet holds 10s of video information.
 The packets starts at 00:00:00, 00:00:10, and 00:00:20. Also imagine that it
takes 1s (propagation time) for each packet to reach the destination.
The receiver can play back the
packets at 00:00:01, 00:00:11, and
00:00:21.Although there is a 1s
time difference between what the
server sends and what the client
sees on the computer screen.

Characteristic :Jitter
 What happens if the packets arrive with different delays? This
phenomenon is called jitter.
 For example, say the first packet arrives at 00:00:01 (1s delay), the second
arrives at 00:00:15 (5s delay), and the third arrives at 00:00:27 (7s delay).
 If the receiver starts playing the first packet at 00:00:01, it will finish at
00:00:11. However, the next packet has not yet arrived; it arrives 4s later.
There is a gap between the
first and second packets and
between the second and the
third as the video is viewed at
the remote site..

Characteristic: Timestamp
 One solution to jitter is the use of a timestamp. If each packet has a
timestamp that shows the time it was produced relative to the first
(or previous) packet, then the receiver can add this time to the time
at which it starts the playback. In other words, the receiver knows
when each packet is to be played.
 To remove gaps between the packets. Imagine the packets in the
previous example have a timestamp of 0, 10, and 20.
If the receiver starts playing
back the packets at 00:00:08,
00:00:18 and 00:00:28.

Characteristic : Playback Buffer
 To be able to separate the arrival time from the playback time, we need a
buffer to store the data until they are played back.
 The buffer is referred to as a playback buffer.When a session begins (the
first bit of the first packet arrives), the receiver delays playing the data
until a threshold is reached.
 In the previous example, the first bit of the first packet arrives at
00:00:01; the threshold is 7s, and the playback time is 00:00:08.The
replay does not start until the time units of data are equal to the threshold
value.
The threshold is measured in time
units of data.
Data are stored in the buffer at a
possibly variable rate, but they are
extracted and played back at a fixed
rate.

Characteristic : Ordering
 A sequence number to order the packets is needed to handle this situation.
 In addition to time relationship information and timestamps for real-time
traffic, one more feature is needed.We need a sequence number for each
packet.
 The timestamp alone cannot inform the receiver if a packet is lost. For
example, suppose the timestamps are 0, 10, and 20.
 The receiver receives just two packets with timestamps 0 and 20.The
receiver assumes that the packet with timestamp 20 is the second packet,
produced 20s after the first.The receiver has no way of knowing that the
second packet has actually been lost.
 Therefore, now it can be played one after the other in continues manner
without any jitter.

Video conferencing application
 Video conferencing: Using a network, a camera and headset, people can
interact as if they were talking face in a room.
 Applications:
 Conducting interviews
 Holding meetings
 Setting up meetings
 Giving lectures
 There are two types of video conferencing. One is called point-to-point
conferencing, which basically is a communication link between any two
locations.
 Another is multipoint conferencing which is a link between a variety of
locations (more then two).

Video Conferencing
 To support real-time audio/video service such as video conferencing, the
following functionalities are essential
 Multicasting (The traffic can be heavy, and the data are distributed by using
multicasting methods.)
 Translation (A translator is a computer that can change the format of a high-
bandwidth video signal to a lower-quality narrow-bandwidth signal.)
 Mixing (If there is more than one source that can send data at the same time
(as in a video or audio conference), the traffic is made of multiple streams.To
converge the traffic to one stream, data from different sources can be mixed.A
mixer mathematically adds signals coming from different sources to create one
single signal.)
 TCP is unsuitable for interactive traffic.
 Multicast services of IP and use of the transport layer protocol.

Voice Over IP
 On real-time interactive audio/video application: voice over IP, or
Internet telephony.The idea is to use the Internet as a telephone
network with some additional capabilities.
 InternetTelephony: with the increased deployment high speed
(broadband) internet connectivity, a growing numbers of individuals
are using internet for voice telephony.
 Protocols to supportVOIP:
1. Session Internet protocol (SIP)
2. H.323

Session Initiation Protocol (SIP)
 SIP is a text-based protocol, as is HTTP. SIP, like HTTP, uses
messages. Six messages such as:
 INVITE,ACK, BYE, OPTIONS, CANCEL, and REGISTER.
 Address: IPv4/Email/Phone number

Tracking the Callee
 What happens if the callee is not sitting at his/her terminal? He/ She may
be away from his/ her system or at another terminal.
 SIP has a mechanism (similar to one in DNS) that finds the IP address of
the terminal at which the callee is sitting.
DHCP of Callee may
not have permanent
IP. So to track the
callee the register is
used.The register
replays and provides
IP address.

H.323
27
 H.323 is a standard designed by ITU to allow telephones on the public telephone network
to talk to computers (called terminals in H.323) connected to the Internet.
 Gateway transforms a telephone network(PSTN ) message to an Internet message.
 ForAudio H.323 uses protocols: compression code, RTP, RTCP, H.225
 For Control and signaling H.323 uses protocols: Q.931 and H.245
 H.323 Operations
 A terminal sends a message to the gatekeeper, which responds with the IP address.
 The terminal and gatekeeper communicate, using H.225 message to negotiate bandwidth.
 The terminal, gatekeeper, gateway, and telephone communicate by using Q.931 to set up
connection.
 H.245 is used to negotiate the compression method.
 RTP is used for audio exchange and RCTP for management.
 Q.931 to terminate connection.

VOIP Programs
 Programs such as Skype or GoogleTalk,….etc
 Programs property:
 Program is a peer-to-peerVOIP client.
 Programs have become very popular.
 Two people can speak with each other using headsets and microphones
connected to their computers directly.
 It is free between any two computers.
 Programs have used good voice compressor providing very good quality
audio.
 It also supports instant messaging, search and file transfer.
 It is encrypted

Multimedia lecture6

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