Multimedia lecture6


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Multimedia lecture6

  1. 1. Eng: Mohammed Hussein1 Republic of Yemen THAMAR UNIVERSITY Faculty of Computer Science& Information System Lecturer, and Researcher atThamar University By Eng: Mohammed Hussein
  2. 2. 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
  3. 3. 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 3 Eng: Mohammed Hussein
  4. 4. 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: 4 Eng: Mohammed Hussein 1: GET: audio/video file 2: RESPONSE Media player 3: audio/video file Web server Server Browser Client
  5. 5. Streaming stored audio/video  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,.) 5 Eng: Mohammed Hussein 1: GET: metafile 2: RESPONSE Media player 3: metafile Web server Server Browser Client 5: RESPONSE 4: GET: audio/video file
  6. 6. Streaming stored audio/video  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. 6 Eng: Mohammed Hussein 1: GET: metafile 2: RESPONSE Media player 3: metafile Web Server Server Browser Client 5: RESPONSE 4: GET: audio/video file Media Server
  7. 7. 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
  8. 8. VOD Requirements of educational Eng: Mohammed Hussein8  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.
  9. 9. The Distribution Network Eng: Mohammed Hussein9  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.
  10. 10. Media servers Eng: Mohammed Hussein 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)
  11. 11. What is Flash Media Server (FMS)? Eng: Mohammed Hussein11  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
  12. 12. 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
  13. 13. Storage issue related to VOD services Eng: Mohammed Hussein13  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
  14. 14. 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 14 Eng: Mohammed Hussein
  15. 15. Streaming Live Audio/Video Eng: Mohammed Hussein15  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
  16. 16. Real-Time Interactive Audio/Video Eng: Mohammed Hussein16  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
  17. 17. Characteristic : Time Relationship Eng: Mohammed Hussein 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.
  18. 18. Characteristic :Jitter Eng: Mohammed Hussein18  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..
  19. 19. Characteristic: Timestamp Eng: Mohammed Hussein19  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.
  20. 20. Characteristic : Playback Buffer Eng: Mohammed Hussein20  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.
  21. 21. Characteristic : Ordering Eng: Mohammed Hussein21  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.
  22. 22. Video conferencing application Eng: Mohammed Hussein22  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).
  23. 23. Video Conferencing Eng: Mohammed Hussein23  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.
  24. 24. Voice Over IP Eng: Mohammed Hussein24  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
  25. 25. Session Initiation Protocol (SIP) Eng: Mohammed Hussein25  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
  26. 26. Tracking the Callee Eng: Mohammed Hussein26  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.
  27. 27. H.323 Eng: Mohammed Hussein 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.
  28. 28. VOIP Programs Eng: Mohammed Hussein28  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