📝 Scheduling algorithms for QoS (30 MCQs)

📖 Explanation: SIP stands for Session Initiation Protocol, which is the core signaling protocol used to create, modify, and terminate multimedia communication sessions such as voice and video calls over IP networks. Understanding this definition is fundamental for any discussion of VoIP technologies and their operation.

📖 Explanation: The Real-time Transport Control Protocol (RTCP) works alongside RTP to convey statistics, timing, and synchronization data about media streams. By exchanging reports on packet loss, jitter, and delay, RTCP enables endpoints to monitor and adjust the quality of the ongoing transmission, making it essential for synchronized media delivery.

📖 Explanation: Mobility management, such as handling roaming and changing IP addresses, is performed at the Network layer (Layer 3) of the OSI model. This layer is responsible for routing decisions and ensures that devices can maintain connectivity when they move between different wireless access points or subnets.

📖 Explanation: Receiver Reports contain loss statistics that indicate how many RTP packets were not received. SIP implementations can use this feedback to trigger re‑negotiation of codecs or adjust bitrates, thereby adapting the media flow to compensate for the observed degradation.

📖 Explanation: Since SIP signaling over UDP does not guarantee delivery, RTP/RTCP streams, which also use UDP, are subject to the same or greater packet loss under congestion. Consequently, the quality of the media may deteriorate faster than the signaling, requiring additional mechanisms such as retransmission or redundancy.

📖 Explanation: SIP registration includes a priority parameter; the registrar stores multiple contacts with associated priorities. When an incoming INVITE arrives, the server selects the contact with the highest priority, ensuring that calls are routed to the most preferred location according to the user's configuration.

📖 Explanation: An increase in RTT reported by RTCP indicates higher latency on the media path. SIP agents typically use this information to trigger a re‑INVITE promptly, allowing the session to renegotiate parameters such as codec or transport address to better suit the new network conditions.

📖 Explanation: Even when RTP payloads are protected by Secure RTP (SRTP), RTCP packets remain in clear text to allow receivers to report loss, jitter, and delay. A SIP proxy can use these metrics to prioritize or reshape traffic, ensuring that encrypted media still receives appropriate scheduling based on observed quality.

📖 Explanation: SIP is responsible for the signaling aspects: locating participants, negotiating session parameters, and managing call state. RTCP, on the other hand, operates alongside RTP to deliver periodic reports on packet loss, jitter, and delay, thereby monitoring and influencing media quality after the call is established.

📖 Explanation: A SIP registrar records the current location (contact URI) of a user and stores it in a database. A SIP proxy, using that stored information, routes incoming requests to the appropriate contact. Thus, the registrar maintains state while the proxy performs the routing function.

📖 Explanation: When RTP/RTCP is paired with SIP, the RTCP reports supply real‑time measurements of loss, jitter, and delay, which SIP can use to trigger re‑INVITE messages that adjust the bitrate. This feedback loop allows the system to adapt to fluctuating wireless conditions, an ability absent if only SIP signaling is employed.

📖 Explanation: Network‑level scheduling can prioritize packets to reduce variation, but some jitter inevitably remains. A small, adaptive jitter buffer at the receiver smooths residual variation without adding excessive delay. The combination of proactive scheduling and modest buffering typically results in the smallest overall latency.

📖 Explanation: TCP provides built‑in retransmission and ordering, ensuring that SIP messages reach their destination, which is important for establishing and maintaining sessions. However, this reliability incurs additional header overhead and connection‑setup latency. UDP incurs less overhead but lacks guaranteed delivery, making it susceptible to loss in mobile scenarios.

📖 Explanation: When the callee changes location, RTCP reports increased delay and loss, which SIP can use to issue a re‑INVITE that renegotiates codecs or transport addresses. The call‑forwarding logic itself remains unchanged, but the media adaptation driven by RTCP ensures the call continues smoothly despite the new congestion conditions.

📖 Explanation: Weighted‑fair queuing (WFQ) allows the proxy to allocate more bandwidth or lower queuing delay to flows with higher weights, such as emergency calls. This ensures that critical traffic experiences minimal latency, whereas round‑robin cycles through all flows equally and cannot guarantee preferential treatment.

📖 Explanation: RTCP packets are typically sent in clear text, allowing intermediate devices to observe loss, jitter, and delay metrics. While these statistics do not contain payload content, they can be leveraged for traffic analysis, enabling potential attackers to infer call patterns or user locations, thus presenting a privacy concern that must be mitigated through encryption or access control.

📖 Explanation: During call setup, each endpoint includes an SDP (Session Description Protocol) offer within the SIP INVITE that lists its supported media types. The receiving endpoint replies with an answer indicating the common subset. This SDP exchange, not RTCP, decides whether video, audio, or both will be used for the session.

📖 Explanation: Limiting registration to only the necessary user agents and assigning them the minimal set of permissions needed to perform their functions embodies the principle of least privilege. This reduces the attack surface by ensuring that compromised devices cannot gain excessive rights within the SIP infrastructure.

📖 Explanation: Stateless SIP proxies forward messages without retaining any session information, which minimizes processing overhead and latency but limits their ability to make nuanced scheduling choices. Stateful proxies keep track of ongoing calls, allowing them to prioritize or reshape traffic based on current session characteristics, which can be advantageous in mobile environments where conditions change rapidly.

📖 Explanation: In this scenario, the SIP server monitors the user's presence status to know which devices are online. Simultaneously, RTCP receiver reports from each device convey real‑time loss and jitter statistics. By combining both data sources, the proxy can route the call to the device that is both available and currently offering the highest media quality.

📖 Explanation: Creating a dedicated QoS class that assigns higher precedence to SIP signaling packets ensures that call setup and control messages traverse the network before media packets. This approach prevents signaling delays that could otherwise stall call establishment, even when the Wi‑Fi link is heavily loaded with RTP traffic.

📖 Explanation: When a user switches interfaces, RTCP continues to send periodic reports that reflect the new path characteristics (delay, loss). SIP may choose to issue a re‑INVITE based on these reports, but the two mechanisms are not directly coupled; the re‑INVITE is a separate signaling action that can be prompted by, but does not depend on, RTCP feedback.

📖 Explanation: An essential step is to continuously monitor the loss percentage reported in RTCP Receiver Reports and, upon exceeding a predefined threshold, initiate a SIP REFER (or re‑INVITE) to transfer the call to an alternative endpoint that may offer better network conditions.

📖 Explanation: By extending the SIP session timer when RTCP indicates elevated jitter, the system gives the adaptive jitter buffer additional time to smooth out packet timing variations. This prevents the session from being terminated prematurely due to transient network issues, thereby enhancing overall call continuity.

📖 Explanation: The method involves issuing SIP OPTIONS requests to candidate endpoints to verify their availability, while concurrently collecting RTCP delay measurements from each path. The caller then selects the destination with the smallest reported delay, ensuring the chosen route offers the best latency before the actual INVITE is sent.

📖 Explanation: Transport Layer Security (TLS) encrypts SIP signaling, protecting call setup information from eavesdropping, while Secure RTP (SRTP) encrypts the media payload, guaranteeing confidentiality and integrity of the voice/video data. Using both together provides comprehensive protection across the entire communication chain.

📖 Explanation: TCP incorporates built‑in acknowledgment and retransmission mechanisms, ensuring that SIP messages reach their destination even when the underlying wireless link experiences high packet loss. This reliability is critical for call setup and teardown, making TCP a preferable transport for signaling under adverse network conditions.

📖 Explanation: When a UDP‑based SIP INVITE is lost, the sender does not receive a response, leading to a retransmission timeout. After several retries without acknowledgment, the caller concludes that the request has failed and may either abort the attempt or retry using an alternative transport method.

📖 Explanation: Higher jitter indicates greater variability in packet arrival times. To maintain session stability, SIP may shorten its session‑refresh timer, prompting more frequent re‑INVITE or UPDATE messages that can renegotiate buffering parameters and help accommodate the increased jitter.

📖 Explanation: When priority values are identical, SIP registrars typically fall back to the registration timestamp to resolve the tie, selecting the contact that was registered first. This deterministic approach ensures consistent routing behavior when multiple contacts share the same priority.