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📝 Kademlia identifier space (9 MCQs)

📖 From Data Communication and Networks • 29. Peer to Peer Paradigm • 9 questions available

What is Kademlia identifier space?

Kademlia identifier space assigns random 160-bit IDs to nodes and keys, where distance between two IDs is calculated using bitwise XOR, creating a metric space that enables efficient routing based on proximity rather than physical location.

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📝 All Kademlia identifier space MCQs

Q1. RTP sequence numbers are 16‑bit and wrap around after reaching the maximum value. If a sender transmits packets with sequence numbers 65534, 65535, 0, 1, what is the most likely consequence for a receiver that does not handle wrap‑around correctly?

A.The receiver will interpret packet 0 as a duplicate and discard it. ✅
B.The receiver will treat packet 0 as a new packet, causing a gap in the sequence.
C.The receiver will reset its jitter buffer, leading to increased latency.
D.The receiver will ignore the entire stream.
💡 Difficulty: easy | ✅ Correct: A

📖 Explanation: When the receiver assumes a monotonically increasing sequence, the sudden jump from 65535 to 0 appears as a regression. Without wrap‑around logic, it treats the new packet as already received, discarding it. This creates a perceived loss and can disrupt ordering, leading to gaps in the media stream.

Q2. Suppose an ongoing VoIP session loses 5 % of its RTCP packets due to network congestion. Which of the following statements best infers the impact on the sender’s ability to adapt its transmission rate?

A.The sender will receive accurate congestion feedback and reduce bitrate accordingly.
B.The sender will overestimate network quality and maintain or increase bitrate, risking further congestion. ✅
C.RTCP loss does not affect bitrate adaptation because RTP carries all control information.
D.The sender will switch to a lossless codec to compensate.
💡 Difficulty: medium | ✅ Correct: B

📖 Explanation: RTCP provides periodic reports that inform the sender about packet loss, jitter, and round‑trip time. Losing 5 % of these reports means the sender receives incomplete feedback, leading it to assume the network is healthier than it actually is. Consequently, it may keep or raise its bitrate, aggravating congestion and degrading call quality.

Q3. If a stream sends N=10,000N = 10{,}000 RTP packets and the packet loss probability is p=0.02p = 0.02, what is the expected number of lost packets, and how might that affect the jitter buffer size assuming each lost packet requires an additional buffering of 55 ms?

A.Expected loss = 200 packets; jitter buffer must increase by 1 s. ✅
B.Expected loss = 200 packets; jitter buffer must increase by 0.5 s.
C.Expected loss = 200 packets; jitter buffer must increase by 0.1 s.
D.Expected loss = 20 packets; jitter buffer must increase by 0.1 s.
💡 Difficulty: hard | ✅ Correct: A

📖 Explanation: The expected number of lost packets equals N×p=10,000×0.02=200N \times p = 10{,}000 \times 0.02 = 200. If each loss adds 5 ms of buffering, the total extra delay is 200×5 ms=1,000 ms200 \times 5\text{ ms} = 1{,}000\text{ ms} or 1 second. Therefore the jitter buffer must be enlarged by roughly one second to absorb the additional variability.

Q4. Which statement correctly contrasts the primary functions of RTP and RTCP in a multimedia session?

A.RTP carries media data while RTCP carries control information such as participant reports. ✅
B.RTP provides encryption and RTCP provides decryption.
C.RTP handles session initiation and RTCP handles media transport.
D.RTP and RTCP both carry identical media payloads for redundancy.
💡 Difficulty: easy | ✅ Correct: A

📖 Explanation: RTP (Real‑time Transport Protocol) is responsible for the timely delivery of audio, video, or other media streams. RTCP (RTP Control Protocol) operates alongside RTP, exchanging statistics, quality reports, and participant information. This separation allows media to flow continuously while control messages are sent less frequently to monitor and adapt the session.

Q5. Considering scalability in large‑scale deployments, which advantage does SIP have over H.323 that most directly reduces the load on network resources?

A.SIP uses a text‑based protocol that is easier to parse, reducing CPU usage.
B.SIP relies on a distributed registrar architecture, avoiding a single point of failure. ✅
C.SIP integrates media negotiation within the same messages, eliminating the need for separate signaling.
D.SIP supports binary encoding, which halves the bandwidth compared to H.323.
💡 Difficulty: medium | ✅ Correct: B

📖 Explanation: SIP registrars can be deployed in a distributed fashion, allowing multiple servers to share the registration load. This design prevents any single node from becoming a bottleneck, reduces latency for registration queries, and scales more gracefully as the number of endpoints grows, directly easing network resource consumption.

Q6. A video source compresses each frame using a lossy codec that reduces the frame size from 2 MB to 500 KB. If the RTP payload limit is 1 500 bytes, how many RTP packets are required per frame, and what is the approximate bandwidth reduction factor compared to sending the raw frames at 30 fps?

A.≈341 packets; bandwidth reduced by ≈2×.
B.≈341 packets; bandwidth reduced by ≈4×. ✅
C.≈200 packets; bandwidth reduced by ≈4×.
D.≈200 packets; bandwidth reduced by ≈2×.
💡 Difficulty: hard | ✅ Correct: B

📖 Explanation: The compressed frame of 500 KB equals about 512 000 bytes. Dividing by the payload limit of 1 500 bytes yields roughly 341 packets per frame. Raw frames of 2 MB (≈2 097 152 bytes) at 30 fps require about 505 Mbps, whereas compressed frames need about 123 Mbps, giving a reduction factor close to 4×.

Q7. In SIP, each call is identified by a unique Call‑ID header generated from a large identifier space. How does increasing the size of this identifier space most directly affect the probability of Call‑ID collisions in a busy network?

A.It increases the probability because more IDs are generated.
B.It decreases the probability exponentially, making collisions negligible. ✅
C.It has no effect; collisions depend solely on network latency.
D.It causes the Call‑ID to be longer, which slows down processing.
💡 Difficulty: easy | ✅ Correct: B

📖 Explanation: A larger identifier space expands the pool of possible Call‑IDs, making the chance that two independently generated IDs coincide drop dramatically. According to the birthday‑paradox principle, the collision probability falls roughly with the square of the space size, so expanding it renders accidental collisions practically impossible.

Q8. When designing a streaming application that uses RTP together with lossless compression (e.g., FLAC for audio), which trade‑off best describes the relationship between latency, bandwidth, and error resilience?

A.Low latency, high bandwidth, and high error resilience are simultaneously achievable.
B.Low latency requires reduced bandwidth, which compromises error resilience.
C.Lossless compression increases bandwidth demand, potentially raising latency, but improves error resilience because any corruption is detectable. ✅
D.Using lossless compression reduces both bandwidth and latency, but worsens error resilience.
💡 Difficulty: medium | ✅ Correct: C

📖 Explanation: Lossless compression preserves the original data, so any bit error can be detected and often corrected, enhancing error resilience. However, preserving all information typically yields larger file sizes than lossy alternatives, increasing the required bandwidth. The higher bandwidth can introduce additional queuing delay, raising overall latency, which must be managed in real‑time applications.

Q9. If the identifier space for SIP Call‑IDs becomes exhausted due to a bug that reuses IDs too quickly, what mechanism should be introduced at the protocol level to mitigate the risk of session misrouting, and why is this mechanism effective?

A.Introduce a checksum field to verify integrity of Call‑IDs.
B.Add a timestamp component to each Call‑ID, ensuring temporal uniqueness. ✅
C.Require encryption of Call‑IDs to prevent reuse.
D.Limit the number of concurrent calls per user to avoid exhaustion.
💡 Difficulty: hard | ✅ Correct: B

📖 Explanation: Appending a timestamp (e.g., epoch time) to the Call‑ID creates a value that changes with each new session, guaranteeing temporal uniqueness even if the random portion repeats. This prevents accidental reuse of an identifier within the window where a previous session might still be active, thereby avoiding misrouting or cross‑talk.

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