1 / 63

Multimedia Transmission

Multimedia Transmission. Voice and Audio Signals. Voice and other audio signals are analog Must be converted to digital signals to transfer over the network At the destination, digitized voice and audio signals must be converted to analog. Analog-to-Digital Conversion.

forrest-church
Download Presentation

Multimedia Transmission

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. Multimedia Transmission

  2. Voice and Audio Signals • Voice and other audio signals are analog • Must be converted to digital signals to transfer over the network • At the destination, digitized voice and audio signals must be converted to analog

  3. Analog-to-Digital Conversion Done by taking samples of the analog signal's amplitude at specific times The higher the sampling rate and bit rate, the better the sound quality Typical voice sampling frequency is about 11 kHz Sampling frequency of music is about 44 kHz

  4. Jitter High sampling rate requires high bandwidth If sampling rate is too high, some packets will be dropped to accommodate inefficient bandwidth Caused by missing packets and latency

  5. Latency Can be caused by bottlenecks and inefficient equipment Minimal latency can be tolerated for telephone conversation Listener will notice but tolerate a break in data flow of about 250 milliseconds 500 milliseconds is unacceptable to listener

  6. Acoustical Echo Commonly encountered in VoIP when using a laptop and audio headset with a microphone Can be prevented by decreasing the level of amplification Single participant experiencing acoustical echo in a conference call can affect all participants

  7. Simplified Example of ADC and DAC Uses Analog-to-Digital Converter (ADC)and Digital-to-Analog Converter (DAC)

  8. Voice over IP (VoIP) Relies on TCP/IP technology Requires inexpensive equipment Less expensive for long-distance phone calls Also known as Internet telephony

  9. VoIP Transmission Model Uses UDP packets to send voice data Uses TCP protocol for video or audio streaming Can communicate from PC to PC using TCP/IP Can be used by mixing TCP/IP with modern phone technologies such as ISDN (Integrated Services for Digital Network, or older systems such as Public Switched Telephone Network (PSTN)(old twisted pair phone network)

  10. Bandwidth-Shaping Techniques • Relied on by VoIP to ensure quality of service • A bandwidth shaper uses technique that can: • Prioritize network packets by protocol or assigned switch port or port number • Delay the delivery of low-priority packets • Bandwidth rate for switch port can be manually set • Bandwidth can be limited for particular IP addresses, type of protocol, or particular application

  11. Telephone Gateway and H.323 • Telephone gateway connects a packet-style network communications system to a telephone system • Uses the H.323 protocol • H.323 protocol controls processes such as: • Name to telephone number conversion • Call forwarding • Caller ID • Call blocking • Conversion from TCP/IP to wireless • Conversion to European telephone systems

  12. Telephone Gateway Configuration

  13. Quality of Service (QoS) Developed to minimize latency Windows operating systems later than Windows XP install QoS Packet Scheduler by default Works for wired and wireless networks

  14. Session Initiation Protocol (SIP) Does not carry data Rather it uses call signaling, which controls session until session is terminated Skype software is example of SIP application Skype supports telephone communications across Internet using a; soft phone( a virtual telphonic service) or hard phone( a physical telphonic service)

  15. Skype Soft Phone Example

  16. Real-Time Transport Protocol (RTP) • Uses UDP to deliver packets as fast as possible • Does not guarantee packet delivery • Used in video conferencing and gaming • Used with Real-time Transport Control Protocol (RTCP), which: • Ensures quality of service • Monitors performance • Can also be used with SIP, H.255, and H.245

  17. VoIP Troubleshooting • Procedure for troubleshooting depends on: • Type of system installed • Topology • Symptoms of the problem • Questions to ask • Is problem performance related (quality of VoIP service)? • Is problem due to a complete failure of the system (connection failure between destination and source)?

  18. Protocol Analyzer • A protocol analyzer can be used to: • Verify a complete network path • Detect latency • Detect missing packets

  19. Improving VoIP Quality • Routers, switches, and gateways allow for QoS configuration • QoS can be configured to give priority to the packets of a specific application, such as the following • Skype • Online gaming • MSN Messenger • Yahoo! Messenger

  20. Firewall and Blocked VoIP Packets One of most common sources of VoIP problems is computer firewall blocking the packets Check if firewall is configured to allow the port assignments associated with the VoIP software application

  21. Audio Device Configuration Failure can be due to the microphone, speaker, or headset Ensure they are turned on Speaker and microphone configuration and testing can be completed through Windows Control Panel/Hardware and Sound

  22. Impedance Mismatch For example, connecting a two-wire local loop telephone circuit to a four-wire telephone circuit One of the most common causes of VoIP echo Also occurs when VoIP technology uses telephone transmission cables for part of the circuit

  23. Windows 7 Hardware and Sound Menu

  24. Applied Networking You’ve installed a VoIP application on a networked computer. The VoIP application can successfully make calls, but it cannot receive incoming calls. What might be the problem, and how would you fix it?

  25. Applied Networking Your VoIP application successfully connects. However, you can hear the person you called but that person cannot hear you. What might be the problem with your VoIP connection?

  26. Review An analog signal is converted into digital code by taking samples of the analog signal’s amplitude. The number of times the sample is taken during a specific period is referred to as the _____. A. bit rate B. sampling frequency C. sampling rate C. sampling rate

  27. Review An analog signal is converted into digital code by taking samples of the analog signal’s amplitude. How often the amplitude is measured per second is referred to as the _____. A. bit rate B. sampling frequency C. sampling rate B. sampling frequency

  28. Review The number of bits used to represent the amplitude of the analog signal is referred to as _____. A. bit rate B. sampling frequency C. sampling rate A. bit rate

  29. Review The small staggers or hesitations in the delivery sequence of audio or video data is referred to as _____. jitter

  30. Review Jitter is caused by _____ or missing packets. latency

  31. Review The delay of data as it travels to its destination is called _____. latency

  32. Review When a microphone and speaker are in close proximity or the audio is improperly adjusted, _____ occurs. acoustical echo

  33. Review The _____ standard compresses video by predicting which areas of the next frame will change and which areas will not. Areas that do not change are not transmitted or stored. A. ADC B. DAC C. video resolution D. MPEG D. MPEG

  34. Review A(n) _____ can be software, hardware, or both that compresses and decompresses video and audio. codec

  35. Review The _____ protocol uses analog signals to transmit data at a maximum rate of 56 kbps. A. ATM B. Frame Relay C. VoIP D. X.25 D. X.25

  36. Review The _____ protocol is a packet switching protocol that has a maximum data rate of 1.544 Mbps. A. ATM B. Frame Relay C. VoIP D. X.25 B. Frame Relay

  37. Review The _____ protocol divides text and audio/video into cells of 53 bytes each. The cells are placed in sequence, giving higher priority to audio/video cells. A. ATM B. Frame Relay C. VoIP D. X.25 A. ATM

  38. Review The ATM protocol uses _____ Bit Rate for applications such as video conferencing and telephone communications. Constant

  39. Review The maximum size of an ATM cell is 53 bytes, with a maximum payload of _____ bytes. 48

  40. Review VoIP uses _____ packets for voice data and _____ packets for video and audio. UDP, TCP

  41. Review The _____ protocol relies on existing TCP/IP technology and network equipment to deliver voice, audio, video, and multimedia. A. ATM B. Frame Relay C. VoIP D. X.25 C. VoIP

  42. Review A(n) _____ is a technique that prioritizes network packets by protocol or assigned switch port or port number and delays the delivery of low-priority packets. bandwidth shaper

  43. Review A(n) _____ connects a packet-style network communications system to a telephone system using the H.323 protocol. A. telephone gateway B. bandwidth shaper C. ATM switch D. PSTN A. telephone gateway

  44. Review The _____ protocol gives time-sensitive packets a higher priority than data packets. A. H.323 B. SIP C. RTP D. QoS D. QoS

  45. Review The _____ protocol is used to stream voice and video in real time. A. H.323 B. SIP C. RTP D. QoS C. RTP

  46. Review The _____ protocol is used for initiating, maintaining, and terminating the exchange of voice, multimedia, gaming, and chat. A. H.323 B. SIP C. RTP D. QoS B. SIP

  47. Review The _____ protocol controls processes such as name to telephone number conversion, call forwarding, caller ID, call blocking, conversion from TCP/IP to wireless, and conversion to European telephone systems. A. H.323 B. SIP C. RTP D. QoS A. H.323

  48. Review The Skype software program is an example of a(n) _____ application. A. H.323 B. SIP C. RTP D. QoS B. SIP

  49. Review What is one of the most common sources of VoIP problems? A. Incomplete path between source and destination B. Latency C. Computer firewall blocking the packets D. Jitter C. Computer firewall blocking the packets

  50. Review A(n) _____ can be used to verify a complete network path, detect latency, and detect missing packets. protocol analyzer

More Related