📝 Multimedia applications in internet (7 MCQs)
📖 From Data Communication and Networks • 28. Multimedia • 7 questions available
What is Multimedia applications in internet?
Multimedia applications in internet encompass services like video conferencing, streaming entertainment, online gaming, and VoIP that integrate audio, video, and data over IP networks, requiring specialized protocols and quality-of-service mechanisms to handle real-time delivery, synchronization, and user interaction despite variable network conditions and bandwidth constraints.
📝 All Multimedia applications in internet MCQs
Q1. A streaming live audio/video service begins to experience high latency. Which protocol’s behavior most directly explains this symptom?
📖 Explanation: RTCP periodically sends reception reports that contain timing and loss information. When network congestion causes these reports to be delayed, the sender perceives higher end‑to‑end latency. The delay is a direct symptom of RTCP feedback lag, whereas RTP loss or SIP/H.323 issues would manifest differently.
Q2. Which multimedia category is uniquely characterized by requiring bidirectional media flow between participants?
📖 Explanation: Real‑time interactive audio/video supports two‑way communication where each participant can simultaneously send and receive media, enabling conversations, collaboration, or gaming. Stored streaming delivers pre‑recorded content in one direction only, and live streaming, while it may be one‑way, does not inherently require simultaneous return media.
Q3. A video conference must allow participants to join and leave dynamically while maintaining media quality. Which set of protocols best supports this scenario?
📖 Explanation: RTP carries the media, RTCP supplies quality feedback and synchronization, and SIP handles signaling for session initiation, modification, and termination, allowing participants to be added or removed on‑the‑fly. H.323 can also manage sessions but is less flexible for dynamic changes, while UDP/TCP combinations lack built‑in signaling.
Q4. During a streaming session, RTCP reports a packet‑loss rate of greater than 10 %. What should the sender most appropriately infer?
📖 Explanation: A loss percentage above 10 % indicates that the network cannot sustain the current bitrate. The sender should lower the encoding bitrate, reducing packet size and frequency, which helps to match the available bandwidth and lowers further loss, preserving playback continuity.
Q5. Compare streaming stored audio/video with real‑time interactive audio/video for a latency‑sensitive online game. Which statement accurately reflects their trade‑offs?
📖 Explanation: Real‑time interactive streams are designed for minimal delay, which is essential for gaming, but they demand higher bandwidth and precise timing to keep all participants synchronized. Stored streaming can buffer content, reducing latency requirements but cannot provide the immediate responsiveness needed for interactive gameplay.
Q6. In a SIP‑initiated call that uses RTP for media transport, why is RTCP essential?
📖 Explanation: RTCP works alongside RTP by sending periodic control packets that contain statistics on packet loss, jitter, and delay. These metrics enable the sender to adjust encoding parameters, maintain lip‑sync, and ensure a consistent user experience. SIP manages session setup, but without RTCP the media flow would lack adaptive quality control.
Q7. Which protocol is primarily responsible for synchronizing audio and video streams in multimedia sessions?
📖 Explanation: RTP (Real‑time Transport Protocol) carries the media payload and includes timestamps that allow receivers to align audio and video frames correctly. While RTCP provides feedback on delivery quality, the actual synchronization of streams is achieved by RTP’s timing information, making it the core protocol for media alignment.