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πŸ“ DS field in DiffServ (10 MCQs)

πŸ“– From Data Communication and Networks β€’ 30. Quality of Service β€’ 10 questions available

What is DS field in DiffServ?

The DS field is a six-bit segment in the IP header formerly known as the Type of Service byte that carries the Differentiated Services Code Point value, which tells routers how to classify and handle each packet according to predefined service behaviors.

3
Easy
5
Medium
2
Hard

πŸ“ All DS field in DiffServ MCQs

Q1. If a terminal broadcasts a request and receives an IP address from the gatekeeper, what can be deduced about the gatekeeper's role in the SIP environment?

A.It acts as a DNS resolver
B.It functions as a registrar server βœ…
C.It provides media streaming services
D.It encrypts SIP messages
πŸ’‘ Difficulty: easy | βœ… Correct: B

πŸ“– Explanation: The gatekeeper supplies its own address so the terminal can register with it. In SIP this behavior corresponds to the registrar function, which maintains user location information. Hence the logical inference is that the gatekeeper is acting as a registrar server, not as a DNS resolver or media provider.

Q2. When a terminal does not receive a bandwidth allocation response via H.225, which outcome is most likely during the subsequent Q.931 negotiation phase?

A.The negotiation will continue with default bandwidth
B.The call will be immediately terminated
C.The terminal will retry the H.225 request
D.The Q.931 negotiation will stall until bandwidth is allocated βœ…
πŸ’‘ Difficulty: medium | βœ… Correct: D

πŸ“– Explanation: H.225 is responsible for negotiating bandwidth before the call can be set up. If the terminal never receives a bandwidth allocation, the Q.931 layer lacks the needed resource information, causing the negotiation to pause. The call cannot progress until the gatekeeper provides the missing allocation, making stalling the most probable result.

Q3. The SDP fields contain start time 29237218542923721854 and end time 29237254542923725454. What is the session duration, and does it exceed the typical SIP timeout of 3600 seconds?

A.1800 seconds, does not exceed
B.3600 seconds, exactly matches βœ…
C.5400 seconds, exceeds
D.7200 seconds, exceeds
πŸ’‘ Difficulty: hard | βœ… Correct: B

πŸ“– Explanation: Subtracting the start time from the end time gives 2923725454βˆ’2923721854=36002923725454-2923721854 = 3600 seconds. This duration equals the common SIP session timeout of 3600 seconds, meaning the session does not exceed the timeout limit. Therefore the correct answer is 3600 seconds, which matches the typical timeout value.

Q4. Which statement best captures the difference in protocol completeness between SIP and H.323?

A.SIP includes RTP by default
B.H.323 is only a signalling protocol
C.SIP is only signalling while H.323 includes RTP and RTCP βœ…
D.Both are identical in functionality
πŸ’‘ Difficulty: easy | βœ… Correct: C

πŸ“– Explanation: SIP provides only signalling functions and relies on external protocols such as RTP and RTCP for media transport. In contrast, H.323 defines a full suite that mandates the use of RTP and RTCP along with signalling. Thus the most accurate distinction is that SIP is purely signalling, whereas H.323 encompasses both signalling and media transport protocols.

Q5. What is the primary advantage of using H.245 to negotiate compression methods instead of fixing a static codec for all calls?

A.Dynamic negotiation reduces bandwidth waste βœ…
B.It simplifies implementation by avoiding negotiation
C.It guarantees higher audio quality regardless of network conditions
D.It eliminates the need for RTP altogether
πŸ’‘ Difficulty: medium | βœ… Correct: A

πŸ“– Explanation: H.245 enables endpoints to agree on the most efficient codec based on current network capacity, allowing the system to use lower‑bitrate codecs when bandwidth is limited and higher‑quality codecs when ample bandwidth exists. This adaptability minimizes unnecessary bandwidth consumption compared to a static codec that may be suboptimal for many calls.

Q6. If H.225 and Q.931 messages were combined into a single message, what would be the most likely impact on call setup time and protocol integrity?

A.Call setup would be faster and protocol layering preserved
B.Call setup would be slower but more reliable
C.Latency would decrease but the separation of concerns would be violated βœ…
D.No impact on either latency or protocol design
πŸ’‘ Difficulty: hard | βœ… Correct: C

πŸ“– Explanation: Merging H.225 (bandwidth negotiation) with Q.931 (call control) could reduce the number of round‑trips, lowering latency. However, it would break the layered design principle that keeps bandwidth management separate from call signaling, potentially causing interoperability and maintenance problems.

Q7. In the SIP header, the \Via\ field primarily operates at which OSI layer?

A.Application βœ…
B.Transport
C.Network
D.Data Link
πŸ’‘ Difficulty: easy | βœ… Correct: A

πŸ“– Explanation: The \Via\ header records the transport path that a SIP request follows, which is handled by the application‑level protocol. Although it contains transport‑layer information (such as UDP or TCP ports), its primary purpose is to support application‑level routing and loop detection, placing it in the Application layer of the OSI model.

Q8. How do RTP and RTCP interact in an H.323 call to ensure quality of service?

A.RTCP replaces RTP for media delivery
B.RTP carries media while RTCP monitors QoS βœ…
C.RTCP encrypts RTP streams
D.RTP handles signaling while RTCP transports audio
πŸ’‘ Difficulty: medium | βœ… Correct: B

πŸ“– Explanation: In H.323, RTP is responsible for the actual transmission of audio or video packets, whereas RTCP periodically sends control packets that convey statistics such as packet loss, jitter, and delay. These reports enable endpoints to adapt transmission parameters, thereby maintaining the desired quality of service during the call.

Q9. If a firewall blocks UDP traffic on port 5060, which strategy allows SIP communication to continue securely?

A.Change the SIP port to 5061
B.Use SIP over UDP on a different port
C.Disable the firewall entirely
D.Use TCP or TLS transport for SIP messages βœ…
πŸ’‘ Difficulty: medium | βœ… Correct: D

πŸ“– Explanation: When UDP on the standard SIP port is blocked, SIP can fall back to TCP, which provides reliable delivery, or to TLS over TCP for encrypted communication. Both alternatives use the same SIP semantics but traverse firewalls that permit TCP traffic, ensuring continued and secure signaling.

Q10. Which audio codec is mentioned as being used by H.323 for compression?

A.G.711
B.G.723.1 βœ…
C.G.729
D.G.726
πŸ’‘ Difficulty: medium | βœ… Correct: B

πŸ“– Explanation: The passage explicitly states that H.323 uses G.71 or G.723.1 for compression. Among the listed options, G.723.1 is the codec identified in the text as part of the H.323 compression suite, making it the correct choice.

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