Test your basic knowledge |

CCNA Voice - 3

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. It has a large header of __ bytes - Size of headers IP/RTP/UDP






2. How many times can a digital to analog conversion in my network and PSTN total network happen - __ speech is less robust conversion






3. Caused by the length a signal must travel via light in fiber or electrical impulse in copper-based networks.






4. Is the amount of time it takes to actually place a bit or byte onto an interface. Its influence on delay is relatively minimal.






5. Time of day of highest calls on circuit focus those on the __






6. The difference between when the packet is expected and when it is actually received is __. - exists only in __ ___ networks






7. When companies merge make sure you configure the __ __






8. PSQM was developed to 'hear' impairments caused by __ and __ and not packet loss or jitter






9. __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






10. Is the variation of packet interarrival time.






11. What do you use to compress RTP header to 2 or 4 bytes






12. This adaptation of jitter buffer __ is effective in compensating for delays.






13. If a packet is lost What is replayed? packet loss is common in data networks but not in __






14. Who scored highest in the MOS






15. __ builds in some reliability to the connectionless __ protocol - enables reliability with out using the overhead of __ - what protocol send multiples of the same packet __






16. RTP consists of a __ __ and a control part the called __






17. You stop Echo in a PSTN by __ __ and is done in __ which is located on each __






18. Voice samples are packed into the___ ___ of a UDP/RTP packet and thus the network header overhead would be __.






19. PCM and ADPCM are examples of __ __






20. __ recommends no more than __ms delay one-way transmission






21. Propagation delay in conjuction with handeling delay can cause noticeable






22. 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






23. 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)






24. ___describes CELP compression that enables __ to be coded into _ Kbps streams;






25. How do you fix queuing delay - you would add






26. ___ ___ are used within Cisco IOS Software to determine what level of ___ - if any - exists within the network.






27. 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.






28. __ is an _ measurement of voice quality and the __ measures the voice






29. What are 3 things you can do if they want to implement a corporate dial plan






30. __ describes __ coding at __ - __ - __ and __ Kbps; you also can interchange ADPCM voice between packet voice and public phone or PBX networks






31. __ the 64 Kbps __ voice coding technique - This is used for the digital voice delivery in the __ or __






32. You hear your own voice in the background ( 4wire)






33. __ __ conceals interarrival packet delay variation.






34. 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.






35. __ protocol that carries audio and video packets - is the standard for transmitting__ __traffic across packet-based networks






36. How does G.729 header go __/__/__






37. RTP uses the __ __ to determine whether the packets are arriving in order - and it uses the __ __ information to determine the interarrival packet time (jitter).






38. To reduce the overall ___ overhead introduced by the 54-byte header - multiple voice samples can be packed into a single Ethernet frame to transmit. Although this can increase the voice ___ - increasing this count can improve the overall voice___ - e






39. __ is not wasting __ when there is no sound to transmit. At least __ % is wasted






40. 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.






41. On a voip link What is the max packet loss






42. Proxy server is not required in an __ based system






43. 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).






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.






45. A __ can trick the human ear into perceiving a higher-quality voice - but a __ cannot be tricked.






46. __ __ replays the last packet received






47. 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.






48. __is characterized as the amount of time it takes for speech to exit the speaker's mouth and reach the listener's ear.






49. __ __ occurs when you have more than one compression/decompression cycle for each phone call and Voice degradation occurs






50. Preferred codec for skype - PC to phone apps - A free speech codec suitable for robust voice communication over IP.