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Test your basic knowledge |
CCNA Voice - 3
Start Test
Study First
Subjects
:
cisco
,
it-skills
,
ccna
Instructions:
Answer 50 questions in 15 minutes.
If you are not ready to take this test, you can
study here
.
Match each statement with the correct term.
Don't refresh. All questions and answers are randomly picked and ordered every time you load a test.
This is a study tool. The 3 wrong answers for each question are randomly chosen from answers to other questions. So, you might find at times the answers obvious, but you will see it re-enforces your understanding as you take the test each time.
1. Is the variation of packet interarrival time.
jitter
sequence number and timestamp
ends - begin - background noise - signal-to-noise threshold
concealment strategy
2. Satellite transmission takes approximately __ ms for a transmission to reach the satellite and another __ ms for it to come back down to Earth. This results in a total delay of __ ms.
PSQM - objective - computer
250 - 250 - 500
Jitter - interarrival - packet-based
tandem compression
3. __ __ replays the last packet received
IP - RTP - UDP
concealment strategy
tandom encoding
forward error correction (FEC)
4. Caused by the length a signal must travel via light in fiber or electrical impulse in copper-based networks.
Propagation delay
payload field - lower
jitter
concealment strategy
5. Echo cancellers are limited by the total amount of time they wait for the reflected speech to be received __ __. A good time to configure is __ms
payload field - lower
echo tail - 25
Serialization delay
tandem compression
6. Time of day of highest calls on circuit focus those on the __
busy house
Jitter - interarrival - packet-based
adaptive differential pulse code modulation (ADPCM) - G.726 - 32
IP - RTP - UDP
7. These techniques can be grouped together generally as __ __ and include variations such as linear predictive coding (LPC) - code excited linear prediction compression (CELP) - and multipulse - multilevel quantization (MP-MLQ).
adaptive differential pulse code modulation (ADPCM) - G.726 - 32
RTP timestamps - jitter
sequence information - time-stamping
source codecs
8. In an unmanaged - congested network - ___delay can add up to two seconds of delay (or result in the packet being dropped). This lengthy period of delay is___ in almost any voice network. delay)
begins - front-end speech clipping.
queuing - unacceptable
RTP is Realtime Transport Protocol - delay sensitive
G.711 PCM
9. __ __ conceals interarrival packet delay variation.
groups payload - qos - combining
IP - RTP - UDP
jitter buffer
G.726 - ADPCM - 40 - 32 - 24 - 16
10. __by default - sends two 10-ms G.729 speech frames in every packet - If you send more __ frames you put in a packet the fewer ___ you require but if you loose a packet the more __ you lose
tandom encoding
G.723.1 speech or audio - 6.3 - 5.3
jitter
PCM - speech - header - information
11. Voice samples are packed into the___ ___ of a UDP/RTP packet and thus the network header overhead would be __.
continuous media - timing reconstruction
queuing - unacceptable
payload field - lower
Handling delay - queuing delay
12. __ __ occurs when you have more than one compression/decompression cycle for each phone call and Voice degradation occurs
tandom encoding
MOS - subjective -1 - 5
compression - decompression
Jitter - interarrival - packet-based
13. What do you use to compress RTP header to 2 or 4 bytes
data part - RTP Control Protocol (RTCP).
forward error correction (FEC)
CRTP
PSQM - objective - computer
14. PCM and ADPCM are examples of __ __
queuing - unacceptable
waveform codecs
person - computer
purchase leased lines - purchase VPN and use access code - OR put voice on data network
15. To make audio and video work with RTP what do you need. 2 things
ITU-T - 150
sequence number and timestamp
VAD - bandwidth - 50
RTP timestamps - jitter
16. When companies merge make sure you configure the __ __
sequence number and timestamp
G.723.1 speech or audio - 6.3 - 5.3
continuous media - timing reconstruction
dial plan
17. Preferred codec for skype - PC to phone apps - A free speech codec suitable for robust voice communication over IP.
tandom encoding
Reliable User Data Protocol (RUDP) - UDP - TCP - RUDP
RTP timestamps - jitter
iLBC
18. 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 __ ___ for congested networks.
adaptive differential pulse code modulation (ADPCM) - G.726 - 32
Serialization delay
Handling delay - queuing delay
groups payload - qos - combining
19. Simplify the router - use H.323 gatekeeper or voice managers to avoid __ __
last packet - voice
dial plan
tandem compression
QoS
20. The difference between when the packet is expected and when it is actually received is __. - exists only in __ ___ networks
Jitter - interarrival - packet-based
40 - 20 - 8 - 12
busy house
PCM - speech - header - information
21. __ recommends no more than __ms delay one-way transmission
data part - RTP Control Protocol (RTCP).
ITU-T - 150
jitter
IP
22. The problem with VAD is detecting when speech __. This is known as__ Usually - the person listening to the speech does not notice this cut off because it did not record and transmit any codec.
begins - front-end speech clipping.
person - computer
busy house
G.723.1 speech or audio - 6.3 - 5.3
23. Another compression method used often is ____ . A commonly used instance of this is ITU-T __ - which encodes using 4-bit samples - giving a transmission rate of __ Kbps.
software - dynamic - RTP
adaptive differential pulse code modulation (ADPCM) - G.726 - 32
G.729 - voice - 8
Jitter - interarrival - packet-based
24. Propagation delay in conjuction with handeling delay can cause noticeable
Serialization delay
groups payload - qos - combining
forward error correction (FEC)
speech degradation
25. The data part of RTP is a thin protocol that provides support for applications with real-time properties - such as __ __ (for example - audio and video) - including __ __ - loss detection - and content identification.
MOS - subjective -1 - 5
2 - G.723.1
continuous media - timing reconstruction
queuing - unacceptable
26. Customers that have almost unlimited bandwidth - however - consider __ a worthwhile mechanism to enhance reliability and voice quality - mechanism makes it more probable that one of the packets will make the journey from sender to receiver
G.711 - PCM - PSTN or PBX(voip networks)
G.726 - ADPCM - 40 - 32 - 24 - 16
forward error correction (FEC)
source codecs
27. RTCP provides support for real-time conferencing of __ of any size within an Internet - does not send __ - offers __and supports __ media streams
queuing - unacceptable
echo tail - 25
groups payload - qos - combining
waveform codecs
28. RTP uses the __ __ to determine whether the packets are arriving in order - and it uses the __ __ information to determine the interarrival packet time (jitter).
Echo
sequence information - time-stamping
queuing - unacceptable
busy house
29. How many times can a digital to analog conversion in my network and PSTN total network happen - __ speech is less robust conversion
Latency
<1%
DNS - IP
7 - compressed
30. On a voip link What is the max packet loss
groups payload - qos - combining
tandem compression
IP
<1%
31. Proxy server is not required in an __ based system
IP
sequence information - time-stamping
forward error correction (FEC)
G.729 - voice - 8
32. ___ ___ are used within Cisco IOS Software to determine what level of ___ - if any - exists within the network.
2 - G.723.1
DNS - IP
payload field - lower
RTP timestamps - jitter
33. How do you fix queuing delay - you would add
last packet - voice
QoS
G.726 - ADPCM - 40 - 32 - 24 - 16
CRTP
34. RTP consists of a __ __ and a control part the called __
data part - RTP Control Protocol (RTCP).
Echo
last packet - voice
Jitter - interarrival - packet-based
35. __ is a __test given to listeners. The listeners give each sample of speech material a rating of __ (bad) to __ (excellent)
last packet - voice
ends - begin - background noise - signal-to-noise threshold
MOS - subjective -1 - 5
PSQM - objective - computer
36. Bob@nextdoorneighbor.com is a __ entry but coverts then to an __ address
compression - decompression
DNS - IP
tandem compression
forward error correction (FEC)
37. Using G.729 - just __ conversions from D/A cause the MOS score to decrease rapidly but __ is less resilient
7 - compressed
sizing
2 - G.723.1
ITU-T - 150
38. This adaptation of jitter buffer __ is effective in compensating for delays.
IP - RTP - UDP
sizing
PCM - speech - header - information
Echo
39. __ Describes a compression technique that you can use to compress ___or other __ signal components of multimedia service at a low bit rate - MP-MLQ provides greater quality at __ Kbps - CELP provides flexibility at __Kbps
sequence information - time-stamping
G.723.1 speech or audio - 6.3 - 5.3
G.711 - PCM - PSTN or PBX(voip networks)
begins - front-end speech clipping.
40. __ builds in some reliability to the connectionless __ protocol - enables reliability with out using the overhead of __ - what protocol send multiples of the same packet __
Reliable User Data Protocol (RUDP) - UDP - TCP - RUDP
MOS - subjective -1 - 5
ITU-T - 150
Jitter - interarrival - packet-based
41. How does G.729 header go __/__/__
IP - RTP - UDP
40 - 20 - 8 - 12
IP
data part - RTP Control Protocol (RTCP).
42. __ describes __ coding at __ - __ - __ and __ Kbps; you also can interchange ADPCM voice between packet voice and public phone or PBX networks
person - computer
G.726 - ADPCM - 40 - 32 - 24 - 16
queuing - unacceptable
7 - compressed
43. PSQM was developed to 'hear' impairments caused by __ and __ and not packet loss or jitter
tandom encoding
G.723.1 speech or audio - 6.3 - 5.3
PCM - speech - header - information
compression - decompression
44. The jitter buffer found within Cisco IOS ___ is considered a __ queue. This queue can grow or shrink exponentially depending on the interarrival time of the __ packets.
250 - 250 - 500
software - dynamic - RTP
2 - G.723.1
7 - compressed
45. If a packet is lost What is replayed? packet loss is common in data networks but not in __
speech degradation
software - dynamic - RTP
250 - 250 - 500
last packet - voice
46. VAD problems in determining when speech __ and __ - and in distinguishing speech from __ __. Its also known as __ __ __ __
jitter
busy house
ITU-T - 150
ends - begin - background noise - signal-to-noise threshold
47. __ is not wasting __ when there is no sound to transmit. At least __ % is wasted
VAD - bandwidth - 50
DNS - IP
payload field - lower
dial plan
48. __ the 64 Kbps __ voice coding technique - This is used for the digital voice delivery in the __ or __
PCM - speech - header - information
G.711 - PCM - PSTN or PBX(voip networks)
250 - 250 - 500
amplitude - hangover
49. You hear your own voice in the background ( 4wire)
echo cancelers - software - DSP
Latency
Echo
busy house
50. VAD detects a drop-off of speech ___ - it waits a fixed amount of time before it stops putting speech frames in packets. This fixed amount of time is known as __ ( when the person stops talking)
250 - 250 - 500
dial plan
amplitude - hangover
iLBC