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I have a few VoIP questions.

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  • #26739 Reply
    dominic arnold
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    These are some questions for a multi-choice test. I have worked over them and i am confident in my ans but would like a second opinion?

    Any help would be great..just pick a question to answer.

    1. In VoIP networks, the assessment of conversational speech quality differs from listening-only speech quality because it also takes into account the impact of
    end-to-end delay �� end-to-end delay and packet loss �� end-to-end delay and jitter �� end-to-end delay, packet loss and jitter

    2. There are two types of subjective listening test – the Mean Opinion Score (MOS) and the Degraded Mean Opinion Score (DMOS). �� For the MOS test, subjects are required to listen only to the degraded speech �� For the MOS test, subjects are required to listen to both the degraded speech and the original (reference) speech �� For the DMOS test, subjects are required to listen only to the degraded speech �� For the DMOS test, subjects are required to listen to both the degraded speech and the original (reference) speech

    3. For perceived speech quality, a late arrival loss due to delay variation (jitter) has �� the same effect as a network packet loss �� more severe effect than a network packet loss �� less severe effect than a network packet loss �� different effect as a network packet loss. The extent of the effect depends on network conditions.

    4. For G.723.1 codec, the frame size is 30 ms. How many samples are there within a speech frame? �� 30 �� 240 �� 120 �� 60

    5. For a given packet loss rate and codec, if the packet size is increased (more speech frames within a packet), �� The perceived speech quality will degrade. �� The perceived speech quality will improve. �� The end-to-end delay will decrease. �� The end-to-end delay will increase.

    6. When audio and video streams are transmitted through VoIP networks,
    �� The audio quality is more sensitive to packet loss and jitter compared to the video quality. �� The video quality is more sensitive to packet loss and jitter compared to the audio quality. �� The audio and video quality have the same sensitivity to packet loss and jitter. �� None of the above, as the quality depends on the network conditions.

    7. Which of the following is (are) correct? �� Subjective speech quality test is not useful, as it is too time consuming and expensive. �� Subjective speech quality is very important, as it offers a benchmark for verifying objective speech quality measurement methods. �� Subjective speech quality measurement is easier, as we can organize 3-5 subjects to carry out a listening test and get a mean opinion score. �� Subjective speech quality measurement is incorrect, as each subject normally gives a different score.

    8. Which of the following is (are) correct? �� Objective speech quality tests try to obtain an objective MOS score, which is similar to subjective MOS score. �� Objective speech quality tests require the two signals that are compared (original and degraded signals) to be time-aligned. �� ITU P.861, PSQM Perceptual Speech Quality Measurement algorithm is suitable for VoIP. �� ITU P.861, PSQM Perceptual Speech Quality Measurement algorithm is not suitable for VoIP.

    9. If end-to-end delay is more than 450ms, it is very difficult to have a conversation. This conclusion applies to what type of networks? �� PSTN networks �� VoIP networks �� Satellite networks �� Any other networks

    10. For a phone-to-phone VoIP connection, the end-to-end speech quality is affected by �� Jitter buffer adjustment �� Echo �� Background noise �� Codec

    11. If PSQM is used to measure end-to-end speech quality in a VoIP network (assuming no end-to-end jitter)
    �� PSQM value will be zero, if there is no IP network impairment (e.g. packet loss and jitter) �� PSQM value will be very low, if there is no IP network impairment (e.g. packet loss and jitter) �� PSQM value will increase, if PSTN transmission delay increases. �� PSQM value will decrease, if PSTN transmission delay increases.

    12. Which of the following is (are) true about Real-time transport protocol (RTP) and Real-time transport control protocol (RTCP) �� RTP and RTCP are on top of UDP �� RTP is on top of UDP, but RTCP is on top of TCP �� RTP and RTCP are both on top of TCP �� RTP is on top of TCP, but RTCP is on top of UDP

    13. Which is (are) correct? �� RTCP can report network packet loss and late arrival loss due to delay variation �� RTCP can only report network packet loss �� RTCP can only report late arrival loss due to delay variation �� RTCP report can differentiate between burst packet loss and random packet loss.

    14. For GSM codecs, the frame size is 160 samples, the payload size for each frame is 33 bytes. Assuming the packet size is 2 frames/packet, which of the following is (are) correct: �� The timestamp will increment by 160 for each RTP packet �� The timestamp will increment by 320 for each RTP packet �� The timestamp will increment by 33 for each RTP packet �� The timestamp will increment by 66 for each RTP packet

    15. RTP/RTCP is very important for VoIP applications, because �� it can guarantee IP network performance �� it can guarantee real-time applications �� it can guarantee perceived Quality of Service �� it can retransmit all lost packets

    16. End-to-end perceived speech quality for VoIP can be measured by �� Segmented signal-to-noise ratio �� Bit error rate �� PSQM/PESQ �� LPC parameters

    17. For conversational speech quality, interactivity mainly depends on �� End-to-end delay �� End-to-end packet loss
    �� End-to-end jitter �� Codec type used

    18. When a jitter buffer is designed, �� a fixed jitter buffer with a large buffer size is preferred. �� a fixed jitter buffer with a small buffer size is preferred. �� adjustment of jitter buffer during a speech silence period is preferred. �� adjustment of jitter buffer during a speech talkspurt is preferred.

    19. Which is the most important performance indicator for VoIP applications �� Mean Opinion Score (MOS) �� Packet loss �� Jitter �� Network delay

    20. For VoIP applications, �� Jitter will cause end-to-end delay �� Jitter will cause further packet loss �� Jitter has no relationship with end-to-end delay or packet loss, as it is a special impairment �� Jitter can be fully compensated by jitter buffer

    #26740 Reply
    Peiman Amini
    Guest

    Hi all

    I am seaching for a kind of relationship or look up table between packet loss ,jitter and delay and the MOS parameter for different codecs,

    Any help would be great,

    Thanks a lot
    Peiman

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