📝 Streaming live audio video protocols (15 MCQs)
📖 From Data Communication and Networks • 28. Multimedia • 15 questions available
What is Streaming live audio video protocols?
Streaming live audio video protocols are specialized communication standards like RTMP, SRT, or HLS that enable real-time capture, encoding, and distribution of ongoing events over networks with minimal latency, supporting features like low-delay transmission, error resilience, and adaptive quality to deliver live broadcasts reliably despite unpredictable internet conditions.
📝 All Streaming live audio video protocols MCQs
Q1. If a live video stream uses a constant bitrate of 4 Mbps and the network experiences a packet loss that reduces effective throughput by 25 %, what is the maximum duration (in seconds) that can be streamed without buffering assuming a buffer size of 10 MB?
📖 Explanation: Effective throughput = 4 Mbps × 0.75 = 3 Mbps = 0.375 MB/s. Buffer of 10 MB divided by 0.375 MB/s gives ≈ 26.67 s. Therefore option B is correct. The calculation shows how reduced bandwidth directly limits uninterrupted playback time.
Q2. Which protocol is primarily used for streaming live audio/video over the internet?
📖 Explanation: The Real-time Transport Protocol (RTP) is designed to deliver audio and video over IP networks with timing information, making it the standard choice for live streaming, unlike HTTP or FTP which are not optimized for real‑time media.
Q3. Given a binary sequence with long runs of zeros, which statement best explains why run‑length coding typically achieves a higher compression ratio than dictionary coding?
📖 Explanation: Run‑length coding captures the count of consecutive zeros in a single code, so long sequences are represented by a short length/value pair. Dictionary coding must store the entire pattern as a dictionary entry, which adds overhead when the pattern is simple, making run‑length more efficient for such data.
Q4. Using a 4‑bit fixed‑length code per digit, a run‑length encoded sequence of 26 bits is stored in 16 bits. What is the compression ratio expressed as a fraction?
📖 Explanation: Compression ratio is defined as original size divided by compressed size. Here the original size is 26 bits and the compressed size is 16 bits, giving a ratio of . This fraction directly reflects the saving achieved.
Q5. If a playback buffer introduces a fixed delay of 2 seconds and network jitter can add up to 0.5 seconds of variance, what is the worst‑case end‑to‑end latency for a live stream?
📖 Explanation: The worst‑case latency occurs when both the buffer delay and the maximum jitter happen together: 2 s + 0.5 s = 2.5 s. This additive effect shows how each component contributes to total latency, making option A correct.
Q6. What does the acronym 'RTSP' stand for?
📖 Explanation: RTSP stands for Real Time Streaming Protocol, which provides control commands such as PLAY, PAUSE, and TEARDOWN for media sessions. It does not transport media itself, distinguishing it from protocols like RTP.
Q7. When deciding between increasing the video bitrate and employing adaptive bitrate streaming for a live event, which statement most accurately reflects the trade‑off?
📖 Explanation: Adaptive bitrate streaming monitors current bandwidth and dynamically selects an appropriate stream quality, preventing buffer underruns that cause stalls. While higher bitrate can improve quality, it risks rebuffering under limited bandwidth, making option B the correct trade‑off description.
Q8. According to the Nyquist theorem, what is the minimum sampling rate required to faithfully reproduce an audio signal with a maximum frequency component of 20 kHz?
📖 Explanation: Nyquist states that the sampling rate must be at least twice the highest frequency present. For a 20 kHz signal, the minimum rate is . Sampling below this causes aliasing, so option C is correct.
Q9. A file of 8 MB is compressed with a ratio of 1.62 (original/compressed). What is the size of the compressed file rounded to two decimal places?
📖 Explanation: Compressed size = original size ÷ compression ratio = 8 MB ÷ 1.62 ≈ 4.938 MB, which rounds to 4.94 MB. This calculation directly applies the definition of compression ratio, confirming option A.
Q10. Which of the following best differentiates RTP from RTSP in a live streaming architecture?
📖 Explanation: RTP is responsible for delivering time‑sensitive media packets, whereas RTSP issues commands such as PLAY or PAUSE to manage the session. This separation of data and control planes makes option A the accurate distinction.
Q11. Why is buffering essential when streaming live audio/video over networks with variable bandwidth?
📖 Explanation: A buffer temporarily holds incoming data, allowing the player to continue playback during brief periods of reduced bandwidth. By absorbing these fluctuations, buffering prevents interruptions, which is the primary reason for its use in live streaming.
Q12. A live stream divides video into 2‑second chunks. If each chunk experiences a network delay of 0.8 seconds, what is the total end‑to‑end delay after three chunks have been transmitted?
📖 Explanation: Each chunk arrives after its duration plus delay: 2 s + 0.8 s = 2.8 s. After three chunks the cumulative delay is . This adds both playback time and network latency, confirming option A.
Q13. Which codec is most commonly employed for low‑latency live video streaming?
📖 Explanation: H.264, especially in its baseline profile, is widely adopted for live streaming due to its balance of compression efficiency and low encoding/decoding latency. While newer codecs like H.265 offer better compression, they typically introduce higher processing delay, making H.264 the preferred choice for low‑latency applications.
Q14. Consider two live streaming setups: (1) a high bitrate stream with a large 5‑second buffer, (2) an adaptive bitrate stream with a small 1‑second buffer. Which setup generally provides lower average latency?
📖 Explanation: Adaptive bitrate streaming can quickly switch to lower‑quality segments when bandwidth drops, allowing a smaller buffer to be sufficient. The reduced buffer size directly shortens the time between capture and playback, yielding lower average latency than a large static buffer, thus option B is correct.
Q15. How does increasing the compression ratio typically affect perceived quality in live streaming?
📖 Explanation: A higher compression ratio means fewer bits are used to represent the same content. When the bit rate drops below the level needed to preserve detail, artifacts appear, lowering perceived quality. Therefore, option B correctly describes the trade‑off between compression efficiency and visual fidelity.