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CCNA Media And Qos Questions

36 questions · Media And Qos topic · All types, answers revealed

1
MCQhard

A Cisco IOS gateway is connected to a WAN circuit where the physical line rate is 10 Mbps but the contracted CIR is only 4 Mbps. Voice quality suffers during bursts because the provider polices traffic above the CIR and drops excess packets. The engineer wants the gateway to smooth outbound traffic so the provider does not discard bursts. Which configuration element should be applied to the WAN interface?

A.A policy-map using the shape average 4000000 command applied outbound on the WAN interface.
B.A class-map matching voice RTP ports with a set dscp ef action applied inbound on the WAN interface.
C.A policy-map using the police 4000000 command applied outbound on the WAN interface.
D.The bandwidth 4000 command under the physical WAN interface configuration.
AnswerA

Traffic shaping buffers and delays excess packets to conform to a configured average rate, which matches a provider policing at the CIR. Setting shape average to 4000000 bits per second on the outbound WAN interface smooths bursts so the policer sees a rate within contract, reducing drops and preserving voice quality across the link.

Why this answer

When a provider polices at a CIR lower than the physical line rate, the enterprise side must pace its own transmissions. Traffic shaping with shape average set to the CIR buffers excess packets and releases them at a conforming rate, so the provider policer does not see bursts. Applying this outbound on the WAN interface protects voice and data from being discarded.

Exam trap

The trap here is confusing policing, which drops excess traffic, with shaping, which delays it to conform to the contracted rate.

2
MCQmedium

A network administrator is troubleshooting one-way audio issues between a Cisco IP Phone and a Cisco TelePresence endpoint. The path traverses a router with a configured class-map. Which command would you use to verify the packet markings on the ingress interface?

A.show class-map configuration
B.show ip rtp statistics
C.show qos interface
D.show policy-map interface
AnswerD

The service-policy is applied on the ingress interface, so show policy-map interface displays per-class matched and marked packet counters there. This confirms whether the class-map's marking (for example DSCP EF for voice) is actually being applied, revealing why one-way audio occurs.

Why this answer

The command `show policy-map interface` displays the actual packet markings and QoS statistics applied to an interface by a policy-map. Since the class-map is referenced within a policy-map, this command shows the number of packets matching each class and the marking actions (e.g., DSCP, CoS) that were applied. This is the only option that directly verifies the markings on the ingress interface.

Exam trap

CLCOR often tests the confusion between configuration verification commands (like `show class-map`) and operational verification commands (like `show policy-map interface`), so candidates must remember that only the latter shows actual packet markings and statistics.

How to eliminate wrong answers

Option A is wrong because `show class-map` only displays the class-map configuration (match criteria) and does not show any packet markings or interface statistics. Option B is wrong because `show ip rtp statistics` provides RTP stream statistics such as jitter and packet loss, but does not show QoS markings. Option C is wrong because `show qos interface` is not a valid Cisco IOS command for verifying packet markings; the correct command is `show policy-map interface`.

3
MCQmedium

A network engineer is deploying a new branch office with a Cisco IOS XE gateway that connects to the service provider via a 100-Mbps Ethernet WAN link. The engineer needs to ensure that interactive voice traffic always receives priority treatment during congestion, while preventing non-real-time traffic from being starved. Which QoS mechanism should be configured on the WAN interface to guarantee strict priority for voice while still servicing other queues?

A.Enable traffic shaping on the WAN interface to smooth out bursts and reduce congestion for all traffic types.
B.Configure weighted random early detection (WRED) on the WAN interface to drop non-voice packets before congestion occurs.
C.Configure a low-latency queue with the priority command inside a policy-map, and apply it to the WAN interface with service-policy output.
D.Apply class-based weighted fair queueing (CBWFQ) with a bandwidth guarantee for the voice class, and rely on the scheduler to prioritize voice.
AnswerC

The priority command in a policy-map creates a low-latency queue (LLQ) that provides strict priority scheduling for voice traffic while policing the priority class to a configured rate. This ensures voice gets absolute priority during congestion, and the policer prevents voice from starving other queues. Applying the policy-map to the WAN interface with service-policy output activates this behavior.

Why this answer

A low-latency queue created with the priority command inside a policy-map provides strict priority scheduling for voice traffic, ensuring that voice packets are always transmitted first when congestion occurs. The priority policer limits the amount of voice traffic that can enter the priority queue, preventing starvation of other queues. This mechanism is the standard way to guarantee voice quality on a WAN interface.

Exam trap

The trap here is assuming that a bandwidth guarantee alone ensures priority treatment, when in fact only the priority command creates a strict priority queue.

4
Multi-Selectmedium

A collaboration engineer is designing QoS for a Cisco Collaboration deployment where IP phones connect to Cisco Catalyst switches and the WAN edge is a Cisco IOS XE router. The engineer wants to ensure that voice bearer traffic is properly marked and prioritized end to end. Which two actions should the engineer take to establish a consistent QoS trust boundary? (Choose two.)

Select 2 answers
A.Configure a priority queue for voice bearer traffic on the WAN egress interface.
B.Configure the access switch port to trust the DSCP or CoS marking from the IP phone.
C.Configure the WAN router to re-mark all traffic entering from the LAN to DSCP 0.
D.Configure the access switch to trust CoS only on the WAN router uplink.
E.Configure the IP phones to mark all traffic, including PC data, as EF.
AnswersA, B

Even with correct markings, voice needs strict priority treatment during congestion to keep delay and jitter low. A low-latency queue on the WAN egress interface, typically matching DSCP EF with a bandwidth guarantee and policer, ensures voice packets are serviced ahead of data so call quality remains acceptable when the link is saturated.

Why this answer

A consistent trust boundary requires the access switch to accept the phone's DSCP or CoS marking so voice keeps its EF value, and the WAN egress interface to provide a strict priority queue for that marked bearer traffic. Together these steps preserve marking from the endpoint and guarantee low latency across the constrained WAN link, which is the foundation of an end-to-end collaboration QoS design.

Exam trap

The trap here is focusing only on marking or only on queuing, when end-to-end voice quality requires both a trusted marking at the access edge and a priority queue at the congestion point.

5
MCQmedium

A collaboration engineer is deploying a new branch where WAN congestion causes jitter on voice calls. The carrier marks all traffic with DSCP 0 on ingress. The engineer wants the branch Cisco IOS router to reclassify voice bearer traffic into the EF class and apply priority queueing before the traffic enters the MPLS WAN. Which configuration approach accomplishes this goal?

A.Configure RSVP between the branch router and the WAN headend to reserve bandwidth for voice bearer traffic.
B.Configure a class-map matching RTP media ports, then a policy-map that uses the priority command, and apply it outbound on the WAN interface with service-policy output.
C.Configure the WAN interface with mls qos trust dscp and enable auto-qos on the interface.
D.Configure a policy-map with a bandwidth guarantee for voice and apply it inbound on the LAN interface.
AnswerB

This is correct because trust is broken at ingress (DSCP 0), so the router must reclassify. A class-map matching the RTP port range identifies voice bearer, the priority command gives it strict-priority LLQ treatment, and applying the policy-map outbound on the WAN interface shapes and queues the traffic in the correct direction toward the congestion point.

Why this answer

Because the carrier resets all packets to DSCP 0, the branch router cannot rely on trust. The engineer must classify voice bearer by RTP port range, mark or treat it as EF, and give it strict-priority queueing with the priority command in a policy-map applied outbound on the WAN interface. This restores low-latency treatment toward the congested MPLS path.

Exam trap

The trap here is assuming that trusting the carrier's DSCP markings is sufficient when the provider has already zeroed them.

6
MCQhard

In a multi-site deployment, you have a requirement to transcode between G.711 and G.729. If the transcoder resources are exhausted, what is the expected outcome for a new call requiring this conversion?

A.The call will use G.711 regardless
B.The call will be rejected
C.The call will proceed with no audio
D.The call will fall back to an MTP
AnswerB

Exhausted transcoder resources leave no Media Termination Point capable of converting G.711 to G.729, so Cisco Unified Communications Manager cannot allocate the required resource for the new call and rejects it with a resource-unavailable cause code. The stem's transcoding constraint is therefore unsatisfiable, making rejection the expected outcome.

Why this answer

When transcoder resources are exhausted, Cisco Unified Communications Manager (CUCM) cannot allocate the necessary DSP resources to convert between G.711 and G.729 codecs. Without a transcoder, the two endpoints cannot negotiate a common codec, so the call is rejected with a cause code indicating resource unavailability (typically 'No Bandwidth' or 'Unavailable').

Exam trap

CLCOR often tests whether candidates confuse transcoders (codec conversion) with MTPs (DTMF/media termination) — they are distinct DSP resources with different functions, and MTPs cannot perform codec transcoding.

How to eliminate wrong answers

Option A is wrong because the call cannot simply 'use G.711' — the G.729-only endpoint does not support G.711, so no common codec exists without transcoding; the call cannot proceed. Option C is wrong because proceeding with no audio would only happen if the codec mismatch were ignored, but CUCM rejects the call before it is established rather than allowing a silent failure. Option D is wrong because an MTP (Media Termination Point) handles DTMF conversion, media flow-through, and supplementary services — it does not perform codec transcoding between G.711 and G.729; that requires a hardware transcoder (DSP) resource.

7
MCQmedium

When using 'RSVP' for Call Admission Control (CAC), what happens if the available bandwidth on the path is less than the requested amount?

A.The call is placed but with a lower quality codec
B.The call is put on hold until bandwidth is available
C.The call proceeds, but marks the packets with Best Effort
D.The call is blocked
AnswerD

RSVP signals the requested bandwidth along the path; when a router finds insufficient unreserved capacity, it returns a PathErr and the admission decision rejects the flow, so the call setup is blocked rather than rerouted or downgraded.

Why this answer

With RSVP-based Call Admission Control, when a call's RSVP PATH/RESV signaling requests more bandwidth than is available on any link along the path, the router rejects the reservation and the call is blocked (denied). RSVP CAC is a deterministic admission control mechanism: if the requested bandwidth cannot be reserved, the call setup fails rather than proceeding with degraded quality. This ensures that admitted calls receive their guaranteed quality of service.

Exam trap

The trap is assuming RSVP CAC degrades or queues calls when bandwidth is insufficient — candidates may confuse it with QoS marking/dropping mechanisms, but RSVP CAC is binary: it either admits or blocks the call.

How to eliminate wrong answers

Option A is wrong because RSVP CAC does not dynamically downgrade the codec — codec selection happens before or during call setup via other mechanisms, and RSVP either admits or rejects the reservation as requested. Option B is wrong because RSVP does not queue or hold calls for later admission; it immediately rejects the reservation request, and the call fails rather than waiting. Option C is wrong because RSVP CAC does not admit the call with Best Effort marking — that would defeat the purpose of admission control; if bandwidth is insufficient, the reservation is rejected outright, not downgraded to Best Effort.

8
Multi-Selectmedium

Which TWO of the following media resource types are supported in CUCM?

Select 2 answers
A.IP Phone
B.Dial-Peer
C.Media Termination Point (MTP)
D.Voice Gateway
E.Transcoder
AnswersC, E

CUCM registers MTP as a media resource for inserting or removing media streams, enabling supplementary services such as DTMF relay and RSVP. It satisfies the stem's requirement for a supported media resource type alongside transcoders, conference bridges and annunciators.

Why this answer

In Cisco Unified Communications Manager (CUCM), media resource types are configured under Media Resources and include annunciators, conference bridges, Media Termination Points (MTPs), music on hold (MOH) servers, and transcoders. Option C, Media Termination Point (MTP), is correct because an MTP is a software or hardware media resource registered to CUCM that provides supplementary services such as DTMF conversion, RSVP agent functionality, and H.323-to-SIP interworking by terminating and reoriginating media streams. Option E, Transcoder, is correct because a transcoder is a media resource that converts one codec (for example, G.711) to another (for example, G.729) so endpoints with mismatched codecs can communicate.

Option A, IP Phone, is incorrect because a phone is an endpoint device, not a media resource type configured under Media Resources. Option B, Dial-Peer, is incorrect because dial peers are configuration elements on IOS voice gateways (used for call routing), not a CUCM media resource type. Option D, Voice Gateway, is incorrect because a voice gateway is a network device that connects CUCM to external networks such as PSTN or other PBXs, not a media resource type within CUCM.

Exam trap

CLCOR often tests whether candidates can distinguish CUCM media resources (MTP, transcoder, conference bridge, annunciator) from device types (phones, gateways) and IOS constructs (dial-peers).

9
MCQeasy

A network administrator is configuring QoS on a Cisco IOS router for a collaboration deployment. The administrator needs to ensure that voice control traffic, such as H.225 and H.245 signaling, is marked with the appropriate DSCP value. According to Cisco best practices, which DSCP value should be used for call signaling traffic?

A.CS3 (24)
B.AF31 (26)
C.EF (46)
D.AF41 (34)
AnswerA

CS3 (Class Selector 3) is the recommended DSCP for call signaling traffic in Cisco collaboration networks. It provides a higher priority than best effort but lower than voice bearer, which is appropriate for signaling. This marking ensures that call setup and teardown messages are prioritized over data traffic but do not compete with voice media. Cisco documentation consistently recommends CS3 for call signaling.

Why this answer

The correct DSCP for call signaling is CS3 (24). This marking ensures that signaling traffic is prioritized above data but below voice bearer, which is essential for maintaining call quality. Voice bearer uses EF (46), while signaling uses CS3.

This separation allows the network to treat real-time voice media with the highest priority and signaling with the next highest, ensuring that call setup and control messages are not delayed during congestion.

Exam trap

The trap here is confusing voice bearer marking (EF) with signaling marking (CS3), or using outdated values like AF31.

10
MCQhard

You are implementing a QoS policy on a WAN edge router. You have a requirement to prioritize video traffic. Which queueing mechanism is recommended by Cisco to minimize jitter and latency for video streams?

A.Weighted Fair Queuing (WFQ)
B.Priority Queuing (PQ) only
C.Random Early Detection (RED)
D.Low Latency Queuing (LLQ)
AnswerD

LLQ assigns video traffic to a strict-priority queue with a configured bandwidth guarantee, so latency-sensitive packets are drained ahead of other classes. This directly minimises jitter and latency, satisfying the stem's video prioritisation requirement, while the policed priority limit prevents starvation of remaining traffic classes.

Why this answer

Low Latency Queuing (LLQ) is the Cisco-recommended mechanism because it combines a strict priority queue for delay-sensitive traffic (voice/video) with CBWFQ for all other classes, ensuring voice gets absolute priority while preventing starvation of other traffic. This minimizes jitter and latency for real-time streams.

Exam trap

CLCOR often tests the distinction between fair-queuing mechanisms (WFQ, CBWFQ) and strict-priority mechanisms (PQ, LLQ) — candidates must remember that only LLQ provides both strict priority for voice and bandwidth guarantees for other classes without starvation.

How to eliminate wrong answers

Option A is wrong because WFQ provides fair bandwidth sharing based on flow weights but offers no strict priority — it cannot guarantee the low jitter voice requires. Option B is wrong because Priority Queuing alone gives strict priority but can starve lower-priority traffic and lacks the class-based bandwidth guarantees of LLQ. Option C is wrong because RED is a congestion-avoidance mechanism that drops packets probabilistically to prevent tail drop — it does not prioritize or queue voice traffic.

11
MCQmedium

You are observing that video traffic is marked as DSCP 0. How can you fix this at the network level without changing the endpoint configuration?

A.Disable traffic shaping
B.Use an ACL to match the traffic and a policy-map to re-mark it
C.Change the interface speed
D.Enable auto-qos on the interface
AnswerB

Re-marking at the network layer via an ACL classifies the DSCP 0 video flow, and the policy-map rewrites its DSCP value on ingress, so endpoints stay untouched. This satisfies the constraint of fixing marking without changing endpoint configuration.

Why this answer

When video endpoints send DSCP 0 (best-effort), the marking must be corrected in the network path. A class-map matches the traffic (e.g., by ACL identifying RTP/Video over the correct port range), and a policy-map with a set dscp command re-marks it to the desired value such as AF41 for interactive video. Applying this via MQC (Modular QoS CLI) on the ingress interface fixes the marking without touching the endpoint configuration.

Exam trap

CLCOR often tests the misconception that auto-qos will fix bad markings — it actually trusts or applies a fixed template, so if the endpoint sends DSCP 0, you need explicit MQC re-marking.

How to eliminate wrong answers

Option A is wrong because disabling traffic shaping removes rate limiting but does nothing to change the DSCP value carried in the packet header. Option C is wrong because changing interface speed affects bandwidth and congestion behavior, not packet marking. Option D is wrong because auto-qos applies a predefined trust-and-mark template based on what the endpoint already sends — if the endpoint sends DSCP 0, auto-qos will not magically upgrade it to a video class; it typically trusts existing markings or applies a fixed template that may not match the desired video DSCP.

12
Multi-Selecthard

Which THREE of the following are key features provided by a QoS policy-map?

Select 3 answers
A.Bandwidth reservation
B.Traffic marking
C.Traffic policing
D.Interface IP address assignment
E.Dynamic routing updates
AnswersA, B, C

Bandwidth reservation guarantees a minimum share of interface capacity to a class using the priority or bandwidth command within a policy-map. It satisfies the stem by being a core policy-map action, ensuring latency-sensitive or business-critical traffic is not starved during congestion.

Why this answer

A QoS policy-map is the component that applies actions to traffic classes, and bandwidth reservation (A) is a core action it can enforce, typically via the bandwidth or priority commands to guarantee a minimum or strict-priority share of link capacity. Traffic marking (B) is also a key policy-map action, using commands like set dscp or set cos to rewrite QoS bits so downstream devices can classify and treat packets consistently. Traffic policing (C) is another standard policy-map action, implemented with the police command to meter flows and drop or re-mark packets that exceed a committed rate.

The remaining options do not belong: interface IP address assignment (D) is a Layer 3 interface configuration task, not a QoS policy-map function, and dynamic routing updates (E) are handled by routing protocols such as OSPF or EIGRP, not by QoS policy-maps.

Exam trap

CLCOR often tests the distinction between QoS policy-map actions and unrelated interface or routing configurations, so candidates may mistakenly include IP address assignment or routing updates as QoS features.

13
MCQhard

You are configuring a Cisco IOS router to perform traffic shaping for voice traffic. Why is traffic shaping often used in conjunction with queuing?

A.To prioritize voice over data traffic
B.To reduce the CPU utilization of the router
C.To prevent interface congestion by smoothing out traffic bursts
D.To increase the total throughput of the interface
AnswerC

Shaping smooths bursts by buffering excess packets and releasing them at the configured rate, so the interface queue is not overwhelmed by sudden spikes. This directly satisfies the stem's voice-traffic constraint: steady pacing keeps delay and jitter low, letting queuing mechanisms schedule voice reliably.

Why this answer

Traffic shaping is used with queuing to prevent interface congestion by smoothing out traffic bursts — it buffers excess traffic and releases it at a configured rate (e.g., the CIR), ensuring the interface does not exceed the contracted bandwidth. Queuing then schedules the shaped traffic according to class priorities, so shaping and queuing work together to manage congestion and protect voice quality.

Exam trap

CLCOR often tests the confusion between shaping (smooths bursts, buffers excess) and policing (drops excess) and between shaping and queuing (rate control vs. priority scheduling) — candidates must remember that shaping prevents congestion by rate-limiting, while queuing prioritizes within that rate.

How to eliminate wrong answers

Option A is wrong because prioritizing voice over data is the role of queuing (e.g., LLQ), not shaping — shaping controls the rate, not the priority. Option B is wrong because traffic shaping actually adds processing overhead (buffering, scheduling) and does not reduce CPU utilization. Option D is wrong because shaping deliberately limits throughput to the shaped rate; it does not increase total interface throughput.

14
MCQeasy

A network engineer is configuring a Cisco IOS router to classify traffic for a QoS policy. The engineer needs to identify packets based on the DSCP value EF that has already been marked by an upstream IP phone. Which configuration element should be used to match this traffic?

A.A route-map with the match ip dscp ef command applied to the interface.
B.A policy-map with the set dscp ef command.
C.A class-map with the match dscp ef command.
D.An access list applied with the ip access-group command on the interface.
AnswerC

A class-map defines the traffic classification criteria, and match dscp ef identifies packets already marked with the Expedited Forwarding DSCP value. This is exactly the tool used to select voice bearer traffic that an upstream phone or switch has marked, so the policy-map can then apply the desired treatment to that class.

Why this answer

Classification for QoS is performed with class-maps, and matching an existing DSCP marking uses match dscp ef. Since the IP phone has already marked voice bearer as EF, the router only needs to identify those packets so the policy-map can apply priority or bandwidth treatment. Policy-maps define actions, not matching criteria.

Exam trap

The trap here is mixing up the role of class-maps, which classify, with policy-maps, which act on classified traffic.

15
Multi-Selecthard

Which THREE of the following are valid reasons to use a Media Termination Point (MTP)?

Select 3 answers
A.Providing transcoding for G.711 to G.729
B.Bridging different signaling protocols
C.Enabling media anchoring for specific features
D.Increasing voice bandwidth
E.DTMF interworking between SIP and H.323
AnswersB, C, E

MTPs can act as a bridge for protocol-specific signaling needs.

Why this answer

Option B is correct because an MTP can bridge different signaling protocols by terminating the signaling and media on each side, allowing calls to interconnect between otherwise incompatible protocol variants. Option C is correct because MTPs enable media anchoring, which is required for features such as call recording, lawful intercept, and certain supplementary services that need the media stream to pass through a controlled point. Option E is correct because MTPs provide DTMF interworking, converting out-of-band DTMF (for example, SIP INFO or H.323) into in-band or RFC 2833 events so that endpoints using different DTMF transport methods can communicate.

Option A is not correct because transcoding between G.711 and G.729 is the function of a transcoder (hardware or software DSP resource), not an MTP. Option D is not correct because an MTP does not increase voice bandwidth; it terminates and relays media, and any bandwidth impact would typically be neutral or negative, not an increase.

Exam trap

CLCOR often tests the confusion between MTP and transcoder functions — candidates incorrectly select transcoding as an MTP role, but MTPs only relay media, they do not change codecs.

16
MCQmedium

Which feature in CUCM is used to block calls when the available bandwidth in a location is exceeded?

A.Gatekeeper
B.Bandwidth Manager
C.Locations CAC
D.RSVP
AnswerC

Locations call admission control totals the bandwidth allocated to phones at a location and rejects new calls once that limit is exceeded, blocking them before they consume WAN capacity. This directly satisfies the stem's requirement to block calls when location bandwidth is exhausted.

Why this answer

Locations CAC (Call Admission Control) is the correct answer because CUCM uses the Locations configuration to define the maximum bandwidth available between locations and to deny new calls when that bandwidth is exhausted. When a call would exceed the configured audio or video bandwidth for a location pair, CUCM rejects the call with a reorder tone or redirects it to a secondary path if configured. This is the native, built-in CAC mechanism in CUCM.

Exam trap

CLCOR often tests the confusion between CUCM Locations CAC (for SCCP/SIP endpoints registered to CUCM) and gatekeeper CAC (for H.323 zones) or RSVP (network-layer reservation), causing candidates to pick the wrong CAC mechanism.

How to eliminate wrong answers

Option A is wrong because a gatekeeper is an H.323 device that provides CAC for H.323-controlled zones, not the CUCM Locations-based mechanism used for SCCP/SIP endpoints registered directly to CUCM. Option B is wrong because 'Bandwidth Manager' is not a CUCM feature — it is a generic term and not the specific CUCM configuration element that enforces location-based CAC. Option D is wrong because RSVP is a resource reservation protocol that provides CAC at the network layer by reserving bandwidth along the path; while CUCM can be configured to use RSVP, the question asks about the feature that blocks calls when location bandwidth is exceeded, which is Locations CAC.

17
Multi-Selecteasy

Which TWO of the following are common audio codecs used in Cisco collaboration?

Select 2 answers
A.JPEG
B.G.711
C.G.729
D.H.264
E.MP3
AnswersB, C

G.711 is a narrowband waveform codec using pulse-code modulation at 64 kbps, delivering toll quality with minimal processing delay. It is the baseline codec in Cisco collaboration, satisfying the requirement for a common audio codec used across Cisco endpoints and gateways.

Why this answer

G.711 (option B) is a correct answer because it is a standard ITU-T narrowband audio codec (64 kbps PCM, using either A-law or mu-law companding) that is universally supported in Cisco collaboration endpoints and gateways for toll-quality voice. G.729 (option C) is also correct because it is a widely used ITU-T compressed audio codec (8 kbps CS-ACELP) commonly deployed in Cisco voice and collaboration environments to conserve bandwidth on WAN links. Option A (JPEG) is incorrect because it is a still-image compression standard, not an audio codec.

Option D (H.264) is incorrect because it is a video compression standard used for video streams, not audio. Option E (MP3) is incorrect because, although it is an audio codec, it is a consumer music format and is not a standard codec used in Cisco collaboration call signaling/media negotiation.

Exam trap

CLCOR often tests the confusion between audio and video codecs, causing candidates to select H.264 or JPEG (video/image codecs) when asked about audio codecs like G.711 and G.729.

18
MCQhard

An administrator needs to enforce a bandwidth limit for video calls at a specific site in CUCM. Which configuration object should be used?

A.Region
B.Location
C.Partition
D.Device Pool
AnswerB

Locations define the maximum audio and video bandwidth allocated between sites or to a device pool. Applying the limit at the Location object enforces call admission control for video, which per-device or region settings cannot achieve.

Why this answer

In Cisco Unified CM, the Location object is the configuration element that enforces call admission control (CAC) by defining the maximum audio and video bandwidth available between locations. When a video call is placed, CUCM subtracts the call's bandwidth from the Location's video bandwidth pool, and if the limit is exceeded, the call is denied or rerouted. This makes Location the correct object for enforcing a per-site bandwidth limit for video calls.

Exam trap

CLCOR often tests the confusion between Region (codec selection) and Location (bandwidth enforcement) — candidates frequently pick Region thinking it controls bandwidth because both are used in call admission.

How to eliminate wrong answers

Option A is wrong because a Region defines which codec is used between groups of devices (codec preference lists) and does not itself enforce a bandwidth cap. Option C is wrong because a Partition is a logical grouping of directory numbers/route patterns used for calling search space dial-plan control, not bandwidth management. Option D is wrong because a Device Pool groups devices for common settings like date/time, region, and SRST reference, but it does not enforce bandwidth limits.

19
MCQhard

Which statement best describes the difference between policing and shaping?

A.Policing drops traffic, whereas shaping buffers it
B.Policing buffers traffic, whereas shaping drops it
C.Both use buffers to manage congestion
D.Both are applied only on ingress interfaces
AnswerA

Policing enforces a rate limit by discarding excess packets immediately, causing retransmissions and TCP throughput collapse. Shaping instead queues and delays outbound traffic to smooth bursts, avoiding drops. The stem's contrast hinges on this drop-versus-buffer mechanism, making the statement accurate.

Why this answer

Policing and shaping both measure traffic against a configured rate, but they react differently when the rate is exceeded. Policing simply drops (or remarks) packets that exceed the committed rate, making it a hard enforcement mechanism typically applied on ingress. Shaping, on the other hand, buffers excess packets and delays their transmission so they are sent later within the configured rate, smoothing bursts rather than discarding them.

Exam trap

CLCOR often tests the buffer-versus-drop distinction between policing and shaping, and the most common error is reversing the two behaviors or assuming both buffer traffic.

How to eliminate wrong answers

Option B is wrong because it reverses the definitions — policing does not buffer traffic and shaping does not drop it; that inversion is the classic distractor. Option C is wrong because only shaping uses buffers (a shaping queue); policing has no buffering mechanism and instead drops or remarks excess packets. Option D is wrong because policing is commonly applied on ingress (inbound) interfaces, while shaping is typically applied on egress (outbound) interfaces — they are not both ingress-only.

20
MCQmedium

You are configuring QoS for a Cisco Unified Communications Manager (CUCM) deployment. You need to ensure that audio media traffic between endpoints is marked correctly. Which DSCP value should be assigned to RTP audio traffic according to the Cisco SRND best practices?

A.DSCP 46 (EF)
B.DSCP 34 (AF41)
C.DSCP 24 (CS3)
D.DSCP 0 (BE)
AnswerA

DSCP 46 maps to Expedited Forwarding, the per-hop behaviour the Cisco SRND specifies for voice RTP media. EF provides low latency, low jitter and guaranteed bandwidth through priority queuing, matching the marking requirement for audio media between endpoints.

Why this answer

Cisco SRND best practices mark voice RTP media with DSCP EF (46), which maps to the Expedited Forwarding per-hop behavior defined in RFC 3246. EF provides low latency, low jitter, and low loss — exactly what voice media requires. Call signaling is marked CS3 (24), and video is typically AF41 (34), so EF is reserved for audio.

Exam trap

CLCOR often tests whether candidates confuse voice (EF 46), video (AF41 34), and signaling (CS3 24) markings — mixing up these three is the most common QoS mistake.

How to eliminate wrong answers

Option B is wrong because DSCP 34 (AF41) is the standard marking for interactive video, not voice audio — using it for voice would put audio in the wrong queue and degrade call quality. Option C is wrong because DSCP 24 (CS3) is used for call signaling (SCCP, SIP) and other control traffic, not for RTP media. Option D is wrong because DSCP 0 (BE) is best-effort with no QoS treatment, which would cause voice quality problems under congestion.

21
MCQeasy

What is the primary function of a 'Class Map' in a Cisco IOS QoS configuration?

A.To categorize and match specific traffic flows
B.To define the action taken on traffic
C.To set the DSCP value for a packet
D.To assign a specific interface for traffic
AnswerA

A class map defines match criteria such as ACLs, DSCP values or protocol ports, identifying which packets belong to a traffic class. It performs classification only; the policy map then applies actions, so the class map's primary function is categorising and matching specific traffic flows.

Why this answer

A Class Map in Cisco IOS QoS is used to identify and classify traffic by matching criteria such as ACLs, DSCP, CoS, or NBAR protocols. It defines 'what' traffic to match, and is later referenced by a Policy Map that defines the action. This separation of classification (class-map) from action (policy-map) is the foundation of Modular QoS CLI (MQC).

Exam trap

CLCOR often tests the class-map vs. policy-map distinction — candidates confuse classification (match) with action (set/police), picking the option that describes policy-map behavior.

How to eliminate wrong answers

Option B is wrong because defining the action taken on traffic is the role of the Policy Map, not the Class Map — the policy-map references a class-map and applies actions like police, shape, or bandwidth. Option C is wrong because setting the DSCP value for a packet is a marking action performed inside a policy-map (via the 'set dscp' command), not a class-map function. Option D is wrong because assigning an interface for traffic is done via service-policy attachment or routing, not by a class-map, which only matches traffic characteristics.

22
MCQeasy

What is the default behavior for DSCP markings on traffic originating from a Cisco IP Phone that has not been configured with specific QoS policies?

A.Marked as DSCP 34 (AF41)
B.Traffic is not marked until a policy-map is applied
C.Marked as DSCP 46 for voice and CS3 for signaling
D.Marked as DSCP 0 (Best Effort)
AnswerC

Absent configured QoS policies, the phone applies its default markings: voice media as DSCP 46 (EF) for low-latency treatment and call signalling as CS3, allowing the network to prioritise voice traffic out of the box.

Why this answer

Cisco IP Phones ship with a built-in QoS trust behavior: voice media (RTP) is marked DSCP 46 (EF) and call signaling (SCCP/SIP) is marked CS3 (DSCP 24) by default, even before any switch or CUCM QoS policy is applied. This allows the phone to self-mark its traffic so the upstream switch can trust it via 'mls qos trust device cisco-phone' or 'trust dscp'. The phone only re-marks if a CUCM softphone/phone security profile or a switch policy overrides it.

Exam trap

CLCOR often tests the assumption that QoS markings require an explicit policy-map on the phone; in reality, Cisco IP Phones self-mark voice as EF (46) and signaling as CS3 by default, and the exam expects you to know those exact values.

How to eliminate wrong answers

Option A is wrong because DSCP 34 (AF41) is the standard marking for interactive video, not voice; voice media uses EF (46). Option B is wrong because the phone marks its own traffic out of the box — no policy-map is required on the phone for default markings to appear. Option D is wrong because DSCP 0 (Best Effort) is the default for untagged/generic host traffic, not for Cisco IP Phone voice and signaling streams, which are pre-marked.

23
MCQhard

An engineer is troubleshooting one-way audio for calls traversing a Cisco IOS XE router with NAT between an internal CUCM cluster and an external SIP provider. Signaling completes, but RTP from the provider never reaches the internal phone. The engineer confirms the provider is sending media to the signaling address. Which technology should be enabled on the router to allow the correct media address to be signaled to the provider?

A.Media Termination Point
B.SIP Normalization and Transparency
C.Application Layer Gateway for SIP
D.RSVP
AnswerC

The SIP ALG inspects the Session Description Protocol inside SIP messages and rewrites embedded connection and media addresses so the public address is signaled instead of the private one. This lets the provider send RTP to the NAT-translated address, restoring two-way audio for the internal phone behind the router.

Why this answer

When SIP signaling crosses a NAT boundary, the SDP body often carries a private IP address that the far end cannot reach, producing signaling success but one-way or no audio. A SIP Application Layer Gateway on the Cisco IOS XE router inspects the SDP and rewrites the connection and media attributes to the translated public address, so the provider sends RTP to the correct destination and audio flows in both directions.

Exam trap

The trap here is assuming that SIP normalization or an MTP solves NAT traversal, when the specific failure is an unreachable private media address embedded in SDP that only an address-rewriting ALG corrects.

24
MCQeasy

Which DSCP value is recommended for Call Signaling traffic in a Cisco Unified Communications network?

A.AF31 (26)
B.EF (46)
C.CS3 (24)
D.AF41 (34)
AnswerC

CS3 (24) is the standard DSCP marking for Call Signaling traffic in Cisco Unified Communications networks, matching the voice deployment model's per-hop behaviour classes. This satisfies the stem's constraint: the recommended DSCP value for Call Signaling.

Why this answer

Cisco recommends DSCP CS3 (24) for call signaling traffic in a Unified Communications network, which includes SCCP, SIP, and H.323 signaling between endpoints and call control. CS3 provides a distinct per-hop behavior from voice bearer traffic, which uses EF (46). This separation lets QoS policies prioritize voice media above signaling while still protecting signaling from best-effort traffic.

Exam trap

The trap is confusing voice bearer marking with call signaling marking: candidates who remember 'voice is EF' pick EF for signaling, but EF is reserved for RTP media, and signaling uses CS3 (24).

How to eliminate wrong answers

Option A is wrong because AF31 (26) is typically used for call signaling in some legacy or alternate QoS designs (and for video signaling in others), but Cisco's current UC SRND recommends CS3 for call signaling, not AF31. Option B is wrong because EF (46) is reserved for voice bearer (RTP media) traffic, which requires the lowest latency and jitter; using it for signaling would compete with voice and violate the intended QoS model. Option D is wrong because AF41 (34) is used for interactive video (video conferencing) bearer traffic, not call signaling.

25
MCQeasy

A network administrator is troubleshooting poor voice quality on a Cisco Unified Communications Manager cluster. The administrator suspects that the issue is related to the Differentiated Services Code Point (DSCP) value being used for voice bearer traffic. Which DSCP value is recommended by Cisco for voice bearer traffic to ensure it receives priority treatment in the network?

A.CS3 (Class Selector 3)
B.AF41 (Assured Forwarding 41)
C.EF (Expedited Forwarding)
D.AF31 (Assured Forwarding 31)
AnswerC

EF (Expedited Forwarding), which corresponds to DSCP 46, is the recommended marking for voice bearer traffic. It indicates that packets require low latency, low jitter, and low loss, and network devices should provide priority queueing for EF-marked traffic. This ensures that voice packets are transmitted first during congestion, maintaining voice quality. Cisco's QoS baseline recommends EF for voice bearer.

Why this answer

Cisco recommends marking voice bearer traffic with EF (Expedited Forwarding), which is DSCP 46. This marking ensures that voice packets are placed in a low-latency queue and receive priority treatment across the network. Other DSCP values like AF31, CS3, and AF41 are used for different traffic types such as call signaling and video, and do not provide the strict priority required for voice.

Exam trap

The trap here is confusing voice bearer marking with call signaling marking, leading to the selection of CS3 or AF31 instead of EF.

26
MCQhard

A collaboration engineer is deploying a new branch office with Cisco Unified Communications Manager and wants to ensure that voice traffic is marked with the appropriate DSCP value. The service provider requires that voice bearer traffic use EF ( Expedited Forwarding). Which DSCP value should be configured on the IP phones for voice media?

A.AF41 (34)
B.CS3 (24)
C.CS5 (40)
D.EF (46)
AnswerD

EF (Expedited Forwarding) is the recommended DSCP for voice bearer traffic. It provides low loss, low latency, low jitter, and assured bandwidth. The service provider expects EF for voice, ensuring proper treatment across the network. Configuring IP phones to mark voice media as EF aligns with industry best practices.

Why this answer

Voice bearer traffic should be marked with EF (Expedited Forwarding, DSCP 46) to ensure low latency, jitter, and loss. This is a standard recommendation and is often required by service providers. The IP phone configuration must set the voice media DSCP to EF, while call signaling typically uses CS3.

Exam trap

The trap here is confusing voice media DSCP with call signaling DSCP; voice media uses EF, while signaling uses CS3.

27
Multi-Selecthard

Which THREE of the following factors impact the quality of video calls in a network?

Select 3 answers
A.Packet loss
B.Interface description
C.Jitter
D.Router uptime
E.Latency
AnswersA, C, E

Packet loss directly degrades video quality: dropped RTP packets cause macroblocking, freezing and audio artefacts, since real-time video cannot retransmit within its latency budget. It is one of the three factors, alongside latency and jitter, that determine call quality.

Why this answer

Packet loss (A) is correct because dropped RTP/UDP packets in a video stream cause visible artifacts, freezing, and audio gaps, directly degrading call quality. Jitter (C) is correct because variation in packet inter-arrival times disrupts the playout buffer, causing choppy audio and stuttering video even when average delay is acceptable. Latency (E) is correct because high end-to-end delay (typically above 150 ms one-way) impairs interactivity and, combined with jitter and loss, degrades the real-time experience.

Interface description (B) is only a text label on an interface and has no effect on traffic handling, and router uptime (D) is an availability metric that does not by itself determine video quality as long as the device is forwarding packets.

Exam trap

The trap is including plausible-sounding but non-QoS factors like interface descriptions or uptime — candidates sometimes pick operational/administrative attributes instead of the three canonical real-time impairments (loss, jitter, latency).

28
MCQmedium

A network engineer is configuring a Cisco IOS router to mark voice bearer traffic with the DSCP value EF (46) and call signaling traffic with CS3 (24). The engineer wants to ensure that both marking operations are applied in a single policy-map on the ingress interface. Which configuration element is required to differentiate the two traffic classes before marking?

A.A class-map that matches on IP precedence 5 for voice and 3 for signaling.
B.A class-map that matches voice bearer and signaling using ACLs or protocol-specific match criteria.
C.A policy-map that uses the 'set ip dscp' command with a table-map to translate incoming markings.
D.A route-map that matches on RTP and SIP ports and sets the DSCP values.
AnswerB

To apply different DSCP markings to voice and signaling, the router must first identify each traffic type. A class-map with match statements such as 'match ip dscp' is not sufficient if traffic is unmarked; instead, matching on UDP port ranges (e.g., 16384-32767 for RTP) or on protocols like SCCP/SIP is needed. This class-map then feeds into the policy-map where 'set dscp ef' and 'set dscp cs3' are applied.

Why this answer

The correct approach is to create a class-map that identifies voice bearer and signaling traffic based on protocol and port numbers, then reference that class-map in a policy-map where DSCP values are set. This ensures that both traffic types are properly marked regardless of their initial markings. Using ACLs or protocol-specific match criteria allows precise differentiation, which is essential for downstream QoS treatment.

Exam trap

The trap here is assuming that matching on existing DSCP or IP precedence is sufficient, but unmarked or differently marked traffic would not be classified correctly.

29
Multi-Selectmedium

A network designer is implementing QoS for a Cisco Collaboration deployment. Which two statements about the trust boundary in a Cisco QoS design are true? (Choose two.)

Select 2 answers
A.All traffic from a PC connected to an IP Phone should be trusted by default.
B.The trust boundary should be placed at the core layer to centralize QoS marking.
C.Cisco IP Phones are typically trusted to mark voice traffic correctly, so the switch port should trust DSCP.
D.The trust boundary is only necessary for voice media, not for call signaling.
E.The trust boundary should be as close to the source of traffic as possible.
AnswersC, E

Cisco IP Phones are designed to mark voice media as EF and signaling as CS3. When connected to a switch, the switch port can be configured to trust the DSCP markings from the phone. This simplifies QoS deployment and is a recommended practice in Cisco collaboration networks.

Why this answer

The trust boundary should be as close to the source as possible, and Cisco IP Phones can be trusted to mark voice and signaling traffic correctly. This ensures that QoS markings are preserved end-to-end and that other devices trust them. Placing the boundary at the core or trusting all PC traffic is not recommended.

Exam trap

The trap here is assuming that the trust boundary should be at the core or that all traffic from a phone port is trusted, but only the phone's traffic should be trusted.

30
MCQmedium

A collaboration engineer is deploying a new Cisco Webex Room Kit in a conference room. The endpoint will send 1080p video at 3 Mbps. The network administrator needs to ensure that the video traffic is marked with the appropriate DSCP value and receives proper treatment in the QoS policy. Which DSCP value should be used for this interactive video traffic?

A.CS4 (32)
B.AF41 (34)
C.AF31 (26)
D.EF (46)
AnswerB

AF41 (Assured Forwarding Class 4, Drop Precedence 1) is the recommended DSCP for interactive video traffic in Cisco collaboration networks. It provides a higher priority than best effort but lower than voice, ensuring that video is prioritized over data while not competing with voice. For a Webex Room Kit sending 1080p video, marking with AF41 ensures appropriate queuing and bandwidth guarantees in the QoS policy.

Why this answer

The correct DSCP for interactive video is AF41 (34). This marking ensures that video traffic is treated with assured forwarding and low drop precedence, providing a balance between priority and bandwidth guarantees. Voice uses EF, signaling uses CS3, and interactive video uses AF41.

This hierarchy ensures that voice gets the highest priority, followed by video, then signaling, then data. Marking video as AF41 allows network devices to apply appropriate queuing and congestion management.

Exam trap

The trap here is confusing video marking with voice marking (EF) or using CS4, which is not the standard for interactive video.

31
MCQmedium

A network engineer is configuring a QoS policy on a Cisco IOS XE router for a collaboration deployment. The engineer needs to ensure that voice traffic is placed in a low-latency queue and that the queue has a strict priority, but also wants to limit voice traffic to 1 Mbps to prevent starvation of other traffic. Which queuing mechanism should be used?

A.Class-Based Weighted Fair Queuing (CBWFQ)
B.First-In, First-Out (FIFO) queuing
C.Weighted Random Early Detection (WRED)
D.Low Latency Queuing (LLQ)
AnswerD

LLQ combines strict priority queuing with a policer. It places voice traffic in a priority queue that is serviced before other queues, ensuring low latency and jitter. The policer limits the amount of traffic that can enter the priority queue, preventing starvation of other queues. Configuring LLQ with a bandwidth limit of 1 Mbps meets the requirement exactly.

Why this answer

Low Latency Queuing (LLQ) is the correct mechanism because it provides a strict priority queue for voice traffic while also allowing a bandwidth limit via the priority command. This ensures that voice packets are serviced first, minimizing latency and jitter, but the policer prevents voice from consuming all bandwidth and starving other traffic. CBWFQ lacks strict priority, WRED is for congestion avoidance, and FIFO is not suitable for real-time traffic.

Exam trap

The trap here is assuming that CBWFQ can provide the same low-latency treatment as LLQ, but CBWFQ only offers weighted fair queuing without strict priority.

32
MCQeasy

A collaboration administrator is configuring a new site in Cisco Unified Communications Manager. The site has a 2 Mbps WAN link, and the administrator wants to prevent more concurrent G.711 calls than the link can support by defining the maximum audio bandwidth allowed. Which configuration element in Cisco Unified Communications Manager should the administrator configure?

A.SIP Profile
B.Device Pool
C.Location
D.Region
AnswerC

A Location defines the total audio and video bandwidth available for calls to and from devices assigned to it. When the aggregate bandwidth of existing calls plus the new call would exceed the configured limit, CUCM rejects the call with a not-enough-bandwidth cause. This is the correct object for capping concurrent calls at a site with a constrained WAN link.

Why this answer

Locations in Cisco Unified Communications Manager are the call admission control mechanism for hub-and-spoke WAN designs, tracking aggregate audio and video bandwidth per site. Assigning devices to a Location with a defined maximum audio bandwidth causes CUCM to deny calls that would exceed the limit, which is precisely what the administrator needs to protect a 2 Mbps link from oversubscription.

Exam trap

The trap here is confusing Regions, which choose codecs between sites, with Locations, which actually count and limit aggregate bandwidth for call admission control.

33
MCQmedium

An administrator is designing a hub-and-spoke collaboration deployment. The hub has a 100 Mbps WAN link shared by many branches, and the engineer needs to ensure that a single branch cannot consume more than its fair share of bandwidth during peak periods. The engineer also wants voice calls to remain prioritized. Which QoS mechanism should be configured on the hub router to achieve both goals?

A.Configure WRED on the WAN interface to drop packets from aggressive branches before they consume too much bandwidth.
B.Configure CBWFQ with a bandwidth guarantee per branch class and add LLQ for voice in the same policy-map.
C.Configure traffic shaping with shape average on each branch class to cap the bandwidth each branch can use.
D.Configure priority queueing only, with voice in the high queue and all branch traffic in the default queue.
AnswerB

Class-based weighted fair queueing gives each branch class a guaranteed minimum bandwidth and shares excess proportionally, which prevents one branch from monopolizing the link. Adding low-latency queueing for voice within the same policy-map ensures voice bearer traffic gets strict-priority treatment, satisfying both the fairness and voice-priority requirements in this hub-and-spoke design.

Why this answer

CBWFQ assigns each branch class a minimum bandwidth guarantee and distributes leftover bandwidth proportionally, preventing any single branch from dominating the shared hub link. Pairing it with LLQ for voice in the same policy-map gives voice strict-priority service, so both fairness among branches and voice quality are maintained during peak utilization.

Exam trap

The trap here is treating priority queueing or shaping alone as sufficient for fairness across many branches.

34
Multi-Selectmedium

Which TWO of the following are considered best practices for implementing QoS in a Cisco collaboration network?

Select 2 answers
A.Mark traffic as close to the source as possible
B.Trust all incoming traffic by default
C.Only mark traffic at the WAN edge
D.Extend the trust boundary to the access layer switch
E.Disable all QoS policies to improve performance
AnswersA, D

Marking at the source lets the originating endpoint or access switch set DSCP values before any congestion or queuing occurs, so downstream devices can classify and prioritise on trusted markings rather than re-inspecting every flow. This satisfies the stem's best-practise requirement by preserving classification integrity end to end.

Why this answer

Option A is correct because Cisco QoS best practice is to classify and mark traffic as close to the source as possible, typically on the endpoint or the first-hop access switch, so that the DSCP value is preserved end-to-end and every downstream device can apply consistent per-hop behavior without re-classifying. Option D is correct because extending the trust boundary to the access layer switch allows that switch to trust the DSCP markings from Cisco IP phones and other endpoints, enforcing classification at the network edge rather than re-marking later. Option B is wrong because trusting all incoming traffic by default lets users or rogue devices set arbitrary DSCP values, which can abuse priority queues and break QoS guarantees.

Option C is wrong because marking only at the WAN edge is too late—traffic has already traversed the LAN unclassified, and the WAN edge cannot reliably distinguish applications after congestion or encryption. Option E is wrong because disabling QoS removes classification, queuing, and policing, which degrades voice and video quality rather than improving performance.

Exam trap

CLCOR often tests whether candidates understand the trust boundary concept — the trap is thinking QoS marking should happen only at the WAN edge or that trusting all traffic simplifies design.

35
Multi-Selecthard

A collaboration engineer is troubleshooting poor video quality on a Cisco Collaboration deployment. The engineer suspects that video traffic is not receiving appropriate QoS treatment and wants to verify the classification and marking on a Cisco IOS router. Which two actions should the engineer take to confirm that video traffic is being handled correctly? (Choose two.)

Select 2 answers
A.Use the show call active voice brief command to inspect the codec negotiated for video calls.
B.Run the show interfaces command to check the interface error counters for video drops.
C.Run the debug ip rsvp command to observe reservation messages for video flows.
D.Use the show class-map command to confirm the match criteria and verify the video class is referencing the correct DSCP or ACL.
E.Run the show policy-map interface command to view per-class packet counts and drop statistics on the WAN interface.
AnswersD, E

The show class-map command displays the match statements configured in each class-map. Verifying that the video class matches the correct DSCP value, such as AF41, or the correct access list confirms that the classification logic itself is correct before relying on interface statistics, which is essential when video is not being placed in the expected queue.

Why this answer

Verifying QoS for video requires checking both the classification logic and the applied policy. The show class-map command confirms the match criteria used to identify video, while show policy-map interface reveals per-class match counts and drops on the interface. Together they confirm that video is classified correctly and treated as intended.

Exam trap

The trap here is assuming that call-level or interface-level commands reveal per-class QoS treatment when only policy and class-map output does.

36
MCQmedium

A network engineer is configuring LLQ on a Cisco IOS router to protect voice traffic. The voice class is assigned a priority of 512 kbps. During periods of congestion, which mechanism ensures that voice packets are not delayed by lower-priority traffic?

A.Weighted fair queueing allocates bandwidth equally among all flows, including voice.
B.Traffic shaping buffers excess voice packets and sends them later.
C.The priority queue is serviced only when other queues are empty, but no policer is applied.
D.The priority queue is serviced before other queues, and a policer drops packets that exceed the configured rate.
AnswerD

LLQ creates a strict priority queue that is always serviced first, ensuring minimal delay for voice. A built-in policer enforces the configured rate, dropping excess packets to prevent the priority queue from starving other queues. This combination provides low latency and jitter for voice while maintaining fairness for other traffic.

Why this answer

LLQ provides a strict priority queue that is serviced before other queues, ensuring voice packets experience minimal delay and jitter. The built-in policer enforces the configured bandwidth limit, dropping excess packets to prevent the priority queue from consuming all bandwidth and starving other traffic. This dual mechanism makes LLQ ideal for real-time voice.

Exam trap

The trap here is assuming that LLQ without a policer is sufficient, but the policer is essential to prevent voice from starving other queues.

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