multimedia transmission
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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. - PowerPoint PPT PresentationTRANSCRIPT
Multimedia Transmission
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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
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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
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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
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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
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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
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Simplified Example of ADC and DAC
Uses Analog-to-Digital Converter (ADC) and Digital-to-Analog Converter (DAC)
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Voice over IP (VoIP)
Relies on TCP/IP technology Requires inexpensive equipment Less expensive for long-distance phone calls Also known as Internet telephony
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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)
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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
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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
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Telephone Gateway Configuration
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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
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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)
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Skype Soft Phone Example
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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
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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)?
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Protocol Analyzer
A protocol analyzer can be used to: Verify a complete network path Detect latency Detect missing packets
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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
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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
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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
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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
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Windows 7 Hardware and Sound Menu
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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?
Applied Networking
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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?
Applied Networking
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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
Review
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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
Review
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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
Review
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The small staggers or hesitations in the delivery sequence of audio or video data is referred to as _____.
jitter
Review
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Jitter is caused by _____ or missing packets.
latency
Review
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The delay of data as it travels to its destination is called _____.
latency
Review
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When a microphone and speaker are in close proximity or the audio is improperly adjusted, _____ occurs.
acoustical echo
Review
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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
Review
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A(n) _____ can be software, hardware, or both that compresses and decompresses video and audio.
codec
Review
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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
Review
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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
Review
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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
Review
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The ATM protocol uses _____ Bit Rate for applications such as video conferencing and telephone communications.
Constant
Review
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The maximum size of an ATM cell is 53 bytes, with a maximum payload of _____ bytes.
48
Review
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VoIP uses _____ packets for voice data and _____ packets for video and audio.
UDP, TCP
Review
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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
Review
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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
Review
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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
Review
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The _____ protocol gives time-sensitive packets a higher priority than data packets.
A. H.323
B. SIP
C. RTP
D. QoS
D. QoS
Review
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The _____ protocol is used to stream voice and video in real time.
A. H.323
B. SIP
C. RTP
D. QoS
C. RTP
Review
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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
Review
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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.323B. SIPC. RTPD. QoS
A. H.323
Review
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The Skype software program is an example of a(n) _____ application.
A. H.323
B. SIP
C. RTP
D. QoS
B. SIP
Review
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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
Review
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A(n) _____ can be used to verify a complete network path, detect latency, and detect missing packets.
protocol analyzer
Review
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Acoustical echo A condition that takes place when a microphone and
a speaker are in close proximity or the audio is improperly adjusted, causing feedback.
Analog-to-Digital Converter (ADC) A computer chip designed to change an analog
signal to a digital signal. Asynchronous Transfer Mode (ATM)
A protocol designed especially for transmitting data, voice, and video.
Glossary
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Available Bit Rate (ABR) An ATM data transfer classification most appropriate
for file transfer. It uses the available bit rate associated with the networking medium. The speed of a file transfer using the ABR classification is affected by the amount of traffic on the network system.
Bandwidth shaper A technique implemented in a network by hardware
or software to control the bandwidth through a particular device such as a gateway, shared Internet connection, network switch, or router.
Glossary
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Bit rate The number of bits used to represent the amplitude
of an analog signal. Codec (compressor/decompressor)
Software, hardware, or combination of software and hardware that compress and decompress video and audio information.
Committed Information Rate (CIR) A guaranteed bandwidth a commercial carrier will
provide a subscriber with.
Glossary
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Constant Bit Rate (CBR) An ATM data transfer classification in which a steady
stream of ATM cells move at a predictable rate. Digital-to-Analog Converter (DAC)
A computer chip designed to change a digital signal to an analog signal.
Frame Relay A packet switching protocol that typically uses leased
lines such as T1 to carry data over long distances.
Glossary
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H.323 A telecommunications standard and protocol used
for audio, video, and data communications using IP or packet-type networks defined by the International Telecommunication Union (ITU).
Hard phone A physical telephonic device.
Impedance mismatch The effect of connecting two different electronic
audio systems that have different electronic characteristics.
Glossary
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Jitter The small staggers or hesitations in the delivery
sequence of audio or video data caused by latency or missing packets.
Latency The amount of time it takes a signal to travel from its
source to its destination. MPEG (Moving Picture Experts Group)
An industry standard that ensures compatibility between different cameras, displays, and other multimedia equipment.
Glossary
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Permanent virtual circuit (PVC) A type of connection that behaves as if it were a
hard-wired connection between the destination and source.
Public Switched Telephone Network (PSTN) The older telephone system that uses twisted pair
cabling and analog signals rather than digital. Quality of Service (QoS)
A protocol that gives time-sensitive packets, such as those carrying telephone conversations, a higher priority than data packets.
Glossary
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Real-time Transport Protocol (RTP) A standard and a protocol used to stream voice and
video in real time. Sampling frequency
The number of times per second that a sample of a signal’s amplitude is taken.
Sampling rate The number of times during a specific period that a
sample of a signal’s amplitude is taken.
Glossary
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Session Initiation Protocol (SIP) A standard and a protocol for initiating, maintaining,
and terminating the exchange of voice, multimedia, gaming, chat, and more.
Soft phone A virtual telephonic device.
Telephone gateway A specialized piece of equipment that connects a
packet-style network communications system to a telephone system using the H.323 protocol.
Glossary
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Unspecified Bit Rate (UBR) An ATM data transfer classification that does not
guarantee any speed or meet requirements of any special application such as multimedia or telephony. This classification is typically applied inside TCP/IP frames.
Glossary
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Variable Bit Rate (VBR) An ATM data transfer classification in which the ATM
cell rate automatically adjusts to support time-sensitive data. It uses multiplexing techniques to provide a minimum Constant Bit Rate for time-sensitive audio and video transmissions while controlling the data rate of not time-sensitive data, such as text or plain e-mail.
Glossary
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Variable Bit Rate-Non Real Time (VBR-NRT) An ATM data transfer classification that allows cells
to move at a variable rate. The rate of movement depends on the type of data contained in each cell.
Variable Bit Rate-Real Time (VBR-RT) An ATM data transfer classification that allows cells
to move at a variable rate depending on the cell’s contents to support real-time audio and video transfers.
Glossary
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Voice over IP (VoIP) An Internet telephony protocol designed for high-
performance data delivery and quality of service. It relies on existing TCP/IP technology and existing TCP/IP networking equipment.
X.25 A protocol that uses analog signals to transmit data
across long distances.
Glossary