N10-009 Question 9
Single answerLayer 4 - TransportA company is rolling out a cloud-based voice and video collaboration platform to several branch offices. Users report that during busy periods, calls connect successfully but audio becomes choppy and video freezes for a few seconds at a time. Network monitoring shows occasional packet loss and variable latency on the WAN links. The application team confirms the platform uses both TCP and UDP at the transport layer. Which protocol should the real-time media streams use to provide the best user experience under these conditions?
- A
TCP, because retransmissions ensure missing voice and video packets are delivered in order
- B
UDP, because low-latency delivery is typically preferred over retransmission delays for real-time traffic
- C
TCP, because connection-oriented sessions automatically eliminate jitter on congested links
- D
UDP, because it guarantees packet delivery while using less bandwidth than TCP
Show answer and explanation
Correct answer: B
Explanation
The best answer is UDP because real-time applications such as voice and video typically prioritize low latency over guaranteed delivery. At Layer 4, TCP and UDP serve different purposes: TCP provides reliable, ordered delivery with acknowledgments and retransmissions, while UDP provides connectionless, low-overhead transport without delivery guarantees. For interactive media, delayed packets are often useless by the time they arrive, so retransmission behavior can reduce quality rather than improve it. This is why technologies such as VoIP and many video conferencing platforms commonly use UDP for media streams, even if TCP may still be used elsewhere in the application stack for signaling, authentication, or session management. This approach is consistent with standard networking best practices and with the transport-layer behavior described in common references such as RFC 768 for UDP and RFC 9293 for TCP.
- A. Incorrect.
Incorrect. TCP provides reliability through sequencing, acknowledgments, flow control, and retransmissions, which is valuable for applications such as file transfers and web traffic. However, for real-time voice and video, waiting for lost packets to be retransmitted often introduces additional delay that degrades call quality more than simply losing a small amount of media traffic. This is a common misconception when people equate reliable delivery with better application performance in all cases.
- B. Correct.
Correct. UDP is commonly used for real-time media streams because it has lower overhead and does not wait for acknowledgments or retransmissions. In voice and video communications, timely delivery is usually more important than perfect delivery. Small amounts of packet loss are often preferable to the extra latency introduced by TCP recovery mechanisms. This aligns with common design practices for VoIP and video conferencing, where transport-layer efficiency supports acceptable quality under variable network conditions.
- C. Incorrect.
Incorrect. TCP is connection-oriented, but that does not mean it eliminates jitter. Jitter is variation in packet arrival times caused by network conditions such as congestion, queuing, and path changes. TCP may even worsen perceived performance for real-time applications because retransmissions and congestion control can add delay. Someone might choose this option by confusing session establishment and reliability with quality-of-service improvements.
- D. Incorrect.
Incorrect. UDP does not guarantee packet delivery, packet order, or error recovery. It is a connectionless transport protocol with minimal overhead. While UDP often uses less overhead than TCP, the statement is wrong because it incorrectly claims guaranteed delivery. This distractor targets the common misunderstanding that a lighter protocol can still provide the same reliability guarantees as TCP.