📝 Real time protocols in computer networks (13 MCQs)
📖 From Data Communication and Networks • 28. Multimedia • 13 questions available
What is Real time protocols in computer networks?
Real time protocols in computer networks are a suite of standards designed specifically for delivering time-sensitive multimedia traffic with bounded latency and jitter, providing mechanisms for packet sequencing, timestamping, payload identification, and feedback reporting that general-purpose transport protocols like TCP cannot offer for live audio and video streams.
📝 All Real time protocols in computer networks MCQs
Q1. What does the 32‑bit identifier assigned to a mixer in RTP streams represent?
📖 Explanation: The 32‑bit identifier attached to the mixer is called the Synchronizing Source (SSRC) identifier. It uniquely labels the stream generated by the mixer, allowing receivers to associate packets with the correct source and to manage synchronization across contributors.
Q2. If packet jitter increases, which component most directly mitigates the effect on playback?
📖 Explanation: A playback buffer stores incoming packets and releases them according to their timestamps. When jitter grows, the buffer can absorb timing variations, smoothing out irregular arrival times before the data is sent to the digital‑to‑analog converter, thus preserving continuous playback.
Q3. Which statement best differentiates the responsibilities of the transport layer versus the application layer in real‑time multimedia?
📖 Explanation: The transport layer (often UDP for RTP) provides end‑to‑end packet delivery, optionally adding reliability features. The application layer is responsible for media‑specific tasks such as encoding, codec selection, timestamp generation, and synchronization, which are essential for real‑time interaction but lie above transport functions.
Q4. Applying QoS concepts, which transport‑protocol feature is most critical for maintaining low latency in interactive video conferencing?
📖 Explanation: Low latency demands that lost packets not be retransmitted, because waiting for a retransmission would introduce unacceptable delay. UDP’s lack of built‑in retransmission allows the application to discard missing frames and continue streaming, preserving the real‑time nature of the conference.
Q5. Given that each contributor has a unique CSRC identifier, what inference can be drawn if two packets in the same RTP stream share the same CSRC value?
📖 Explanation: A CSRC identifier labels the source that contributed the payload. When two packets share the same CSRC, the receiver can infer that they were produced by the same contributor (microphone or camera), regardless of where they appear in the combined RTP stream.
Q6. Evaluate the impact of using a larger playback buffer on jitter tolerance versus end‑to‑end delay.
📖 Explanation: A larger buffer can store more out‑of‑order packets, allowing the system to smooth jitter more effectively. However, the trade‑off is that the first packet must wait longer before being played, so overall end‑to‑end latency grows in direct proportion to the buffer size.
Q7. Synthesize a design where RTP packets are encrypted end‑to‑end. Which protocol combination best ensures confidentiality without compromising timing?
📖 Explanation: Secure Real‑time Transport Protocol (SRTP) provides payload encryption and authentication, while DTLS supplies the key‑exchange handshake over UDP. This combination retains the low‑latency characteristics of UDP and adds confidentiality, making it the preferred solution for real‑time encrypted media.
Q8. If an RTP packet is lost but its sequence number indicates a gap, what deduction can the receiver make about the state of the playback buffer?
📖 Explanation: When a sequence‑number gap is detected, the receiver knows a packet is missing. Real‑time buffers typically insert a short silence or placeholder frame to preserve timing, then continue with the next arriving packet, avoiding stalls while maintaining stream continuity.
Q9. When network bandwidth drops, which adaptive strategy most directly preserves real‑time interaction quality?
📖 Explanation: Reducing video resolution lowers the amount of data per frame while keeping the frame rate stable, which maintains interactivity and smooth motion. This approach balances visual quality with the limited bandwidth, ensuring that participants continue to see and hear each other without noticeable pauses.
Q10. Apply the principle of SSRC collision handling: what action should a receiver take upon detecting two streams with identical SSRC but different payload types?
📖 Explanation: When an SSRC collision is detected, the receiver assumes the streams belong to the same logical source. It selects the payload type that best matches its capabilities (often the higher‑priority type) and continues processing, rather than discarding or merging the streams, which could disrupt synchronization.
Q11. Deduce why RTP does not implement its own transport reliability mechanisms, and how this design choice influences protocol layering?
📖 Explanation: RTP is built on top of UDP, which offers minimal latency but no reliability. By leaving loss detection and recovery to higher‑level mechanisms (e.g., application‑specific concealment), RTP avoids the overhead of retransmissions, preserving real‑time performance and keeping the protocol stack modular.
Q12. Analyze how the presence of multiple microphones (contributors) affects the packetization strategy at the mixer?
📖 Explanation: When several contributors send audio, the mixer combines their samples into interleaved frames, preserving the order and timestamps of each source. Each packet carries the CSRC identifiers of the involved contributors, enabling the receiver to separate and process the individual audio streams if needed.
Q13. Which layer of the TCP/IP model is primarily responsible for converting analog audio signals into digital packets in a real‑time system?
📖 Explanation: The conversion of analog audio into digital samples and the subsequent packaging into RTP packets occurs in the application layer, where media capture, encoding, and packet formation are performed before the data is handed down to the transport and network layers.