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[REV] NSCOM01 FINALS REVIEWER

Tags: #◇Computer Science #◇NSCOM01 #◇Conceptual #◇Reviewer


◇NSCOM01 Comprehensive Finals Reviewer

This reviewer covers all core topics for the Term 2 Finals, including VoIP Basics, Real-Time Protocols (RTP, RTCP, RTSP), SIP, P2P Architecture, BitTorrent, and Skype's VoIP implementation.

◇References & Navigation

CONFIRMATION // 理解度の確認

ACTIVE RECALL VERIFICATION

Select an answer to verify conceptual comprehension and reveal the technical rationale.

Q1

To maintain a seamless VoIP conversation, the end-to-end delay must be kept below:

Q2

The fluctuation in end-to-end delay of consecutive packets caused by variable network congestion is known as:

Q3

How does increasing the playout buffer (qq) affect VoIP performance?

Q4

Interleaving is a loss concealment technique that works by:

Q5

Which of the following is true regarding the Real-Time Transport Protocol (RTP)?

Q6

Which RTP header field allows the receiver to properly synchronize playout and handle jitter?

Q7

What is the primary function of the Real-Time Control Protocol (RTCP)?

Q8

RTSP (Real-Time Streaming Protocol) is primarily designed to:

Q9

In SIP, which server is responsible for routing INVITE messages to locate the callee?

Q10

What message is used by a SIP client upon bootup to bind its current dynamic IP address to its SIP URI?

Q11

If a SIP callee receives an INVITE but does not support the requested media codec, what response should they send?

Q12

In a P2P network distributing a file to NN peers, what happens to the total available upload capacity as NN increases?

Q13

What is the role of a Tracker in the BitTorrent protocol?

Q14

BitTorrent's "Rarest First" strategy is designed to:

Q15

The BitTorrent algorithm that rewards peers for uploading and penalizes leechers is called:

Q16

What is the purpose of Optimistic Unchoking in BitTorrent?

Q17

Classic Skype utilized a hybrid architecture consisting of:

Q18

What is a Skype Supernode?

Q19

How does Skype overcome the issue of NAT Traversal when both callers are behind NAT routers?

Q20

During a talk spurt, if audio is encoded at 64 kbps and packetized every 20 msec, what is the size of the payload chunk in bytes?

[REV] NSCOM01 Finals Reviewer | PROJECT ENGRAM