Card Set Information

2011-07-24 21:52:27

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  1. the time it takes for speech to exit the speakers mouth and reach the listeners ear
  2. caused by the length a signal must travel via light in fiber or electrical impulse in copper based networks
    propagation delay
  3. defines many different causes of delay(actual packetization, compression, and packet switching), and is caused by devices that forward the frame through the network.
    includes queuing delay for congested netoworks
    handeling delay
  4. propagation delay in conjuction with handeling delay can cause noticeable
    speech degradation
  5. cisco uses Digital signal process (DSP) to sample speech using what codec
  6. what does cisco use to keep the overhead low
  7. when packets are held in a queue bc of congestion on an outbound interface
    occurs when more packts are sent out thean the interface can handle
    queuing delay
  8. the amount of time it takes to actually place a bit or byte onto an interface
    its influence on delay is relatively minimal
  9. ITU-T, a delay of no more then
  10. what is the time it takes for a satellite transmission to reach the satellite then back to earth
    • 250ms to
    • 250ms from the satellite to earth
    • Total delay 500ms
  11. how many seconds does queueing delay add
    2 sec
  12. the variation of packet interval time
  13. does jitter occure in packet based or switched based networks
    packet based
  14. the difference between when the packet is expected and when it actually recieved
  15. whick conceals interarrival packet delay variable
    -This is bc voice packets have high variable packet interarrival times
    -count late packets
    -adjust target to allow late-packet ratio

    we have to be under 150ms to have this work
    jitter buffer
  16. used within the cisco ios software to determine of jitter exists within the network
    RTP timestamps
  17. the jitter buffer found in the cisco ios queue can grow or shrink exponentially depending on the interarrival time of the RTP packets
    dynamic queue
  18. when analog transmission is passed through amplifers to boost the signal not only was the voice boosed so was
    line noise was amplified
  19. Cisco IOS, by default, sends two 10-ms G.729 speech frames in every packet.
    packetizing pulse code modulation
  20. the more speech frames you put into a packet the
    fewer headers you require (but loose more info)
  21. how do you reduce the IP/RTP/UDP overhead, multiple voice samples can be packed into a
    single eathernet frame to transmit
  22. what layer is udp
    layer 4
  23. more voice samples per frame can what, only if bandwidthih is constrained
    improve voice quality
  24. voice samples per frame and bandwidth utilization impack packet loss, the bigger the value the
    banwidth utilization increases b/c more samples are in the payload field of the UDP/RDP packet
  25. so what kind of packet impack can occure
    larger packet loss
  26. g.729 max voice samples and default
    • 64
    • 2
  27. what is the codec of adaptive differential paulse code modulation (ADPCM)
  28. how many bit samples do we use giving a transmission rate of 32 kbps, ADPCM
    4 bit samples
  29. compression techniques that exploit redudent characteristics of the waveform itsself
    waveform codecs
  30. techniques employ signal processing procedures that compress speech by sending only simplified parametric information about the original speech excitation and vocal tract shaping, requiring less bandwidth to transmit that information.
    source codecs
  31. 64 kbps PCM coding
  32. 40,32, 24, 16 kbps ADPCM coding
    used between voice, public phone or PBX networks
  33. 16Kbps low delay CELP voice compression
  34. when you have bandwidth constrate it can go to 8 kbps
    ok quality
  35. low bandwidth compression technique
    5.3 and 6.3kbps
    good quality but not as good as pbx
    flexibilty when you have low bandwith situations
    based on CELP
  36. free speach codec
    good for voip
    slow quality degradation
    high rebustness to packet loss
    13.33 kbps encoding frame length
    15.20 kbps encoding frame length
    iLBC (Internet Low Bitrate Codec)
  37. used for skype and pc to phone application
  38. test a group of listners on voice quality
    MOS (mean opinion score)
  39. what is the high and low MOS
    1 bad 5 excelent
  40. what is the best sounding codec at 64 kbps
    G.711 PCM
  41. second best codec at 32kbps
    G.726 ADPCM
  42. was developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.
    Perceptual Speech Quality Measurement (PSQM). P.861
  43. for PSQM A person can trick the human ear into perceiving a higher-quality voice, but a
    computer cannot
  44. what you hear your voice in the background
    caused by 4 wire comversion to wire in traditional network
  45. how does the PSTN stop Echo
    echo cancellers
  46. what is echo draw drawbacks
    loud and long
  47. where are the echo cancellers built
    low bit rate codecs on each DSP
  48. where can some maufactures stop echo at
    in software but can be slow
  49. where does cisco voip do all its cancellation at
    DSP or at the transation and codec conversion
  50. 25 ms is good for the user to hear or what will the user think
    that you dropped him
  51. Echo cancellers are limited by the total amount of time they wait for the reflected speech to be received
    echo tail
  52. classify and manage traffic through a data network to keep packet loss to a minimum
  53. the accepted packer loss on a link is
    less than 1%
  54. what device responds if there is periodic packet loss
    voice routers
  55. when you replay the last bit that was not dropped on voice transmission
    concealment strategy
  56. if lost speech is only 20 sec the listener will kntice what the the speach
    not noitice
  57. not wasting bandwidth when there is not sound to transmit
    you can utilize this "wasted" bandwidth for other purposes
    voice activity detection (VAD)
  58. regardless if no one is speaking how mush data is moving
  59. how much of a call is wasted bandwidth
    at least 50 %
  60. when the VAD detects a drop-off of speech amplitude, it waits a fixed amount of time before it stops putting speech frames in packets. This fixed amount of time is known as
    (sending silence)
    hangover (200 ms)
  61. vad is unable to to distinguish between speech and background noise known as
    • singnal to noise threshold
    • (disables vad)
  62. VAD is detecting when speech begins. Typically the beginning of a sentence is cut off or clipped
    front end speech clipping
  63. when a person stops talking not any useable sound
  64. today's toll networks can handle how many a D/A (digital to analog)conversions before voice quality is affected
  65. when using G.729 2 conversations from Digital to Analog will affect what
  66. to tandem encoding. G.729 can handle how many compression/ decompression cycles
  67. Simplify the routerconfiguration
    Use a Cisco IOS Multimedia Conference Manager (for instance, H.323 Gatekeeper).
    Use one of Cisco's management applications
    Thes help you with what
    avoiding tandem compression
  68. audio and video packets
  69. standard for transmitting delay-sensitive traffic across packet-based networks ( audio video)
    Media on demand
  70. RDP uses what to determine whether the packets are arriving in order
    Sequence info
  71. determine delay and jitter
    time stamping info
  72. to determine the interarrival packet time (jitter).
    time stamping and sequence info
  73. a thin protocol that provides support for applications with real-time properties, such as continuous media ( transport of the data)
    data part RTP
  74. for real-time conferencing of groups of any size within an Internet. ( does not send the payload )
    RTP Control Protocol (RTCP).
  75. provides a rich set of data for VoIP management
    RTP Control Protocol Extended Reports (RTCP XR)
  76. header ip 20 udp 12 rtp 8 how much bigger payload then g.729
    2 times
  77. the processing time increse compress header
    decrease it takes to transmite
    slow link throuput is going to increase
    faster link the compression will decrease the throughput
    RTP header compression
  78. enhances without using TCP
    send multiples of the same packet and enable the receiving station to discard the unnecessary or redundant packets.
    Reliable User Data Protocol (RUDP)
  79. mechanism makes it more probable that one of the packets will make the journey from sender to receiver
    forward error correction (FEC)
  80. a worthwhile mechanism to enhance reliability and voice quality, if you have unlimited bandwidth
    forward error correction (FEC)
  81. Purchase leased lines
    Purchase a telephony Virtual Private Network (VPN)
    Take advantage of the existing data infrastructure and put voice on the data network
    how to fix dial plan issues with growing company
  82. use seven diget ext when
    excessive growth
  83. what is compatible with her internet phone codec
  84. H.225 is for
  85. codetic infro is sent (in packet based system)
  86. proxy is not needed in an
    ip bassed system
  87. tjr31@zips.uakron.edu is an example of what
    DNS (domain name system )
  88. converts Bob.nextdoorneighbor.com to a DNS host name and goes to
    DNS server