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CCNA Protocols And Endpoints Questions

20 questions · Protocols And Endpoints · All types, answers revealed

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1
MCQhard

You have a requirement to use secure signaling for SIP endpoints. Which protocol and port combination must be used in the SIP Security Profile?

A.SIP TLS 5061
B.HTTP 443
C.SIP TCP 5060
D.SRTP 5061
AnswerA

SIP TLS on port 5061 encrypts signalling between SIP endpoints, satisfying the requirement for secure signalling. TLS protects headers and credentials that plain SIP on 5060 exposes, and 5061 is the registered port for SIP over TLS. The SIP Security Profile in Cisco Unified Communications Manager expects this pairing.

Why this answer

SIP TLS on port 5061 is the standard secure signaling transport for SIP endpoints in Cisco Unified Communications Manager. The SIP Security Profile in CUCM requires selecting TLS as the transport with port 5061 so that SIP signaling is encrypted with certificates between the endpoint and the call agent.

Exam trap

CLCOR often tests the confusion between signaling encryption (SIP TLS/5061) and media encryption (SRTP), causing candidates to pick SRTP as the 'secure signaling' answer.

How to eliminate wrong answers

Option B is wrong because HTTP 443 is used for web/HTTPS management and XML services, not for SIP signaling. Option C is wrong because SIP TCP 5060 is the unencrypted signaling port and provides no confidentiality or integrity for SIP messages. Option D is wrong because SRTP is the media encryption protocol (typically on even-numbered RTP ports), not a signaling protocol, and it does not run on 5061.

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2
MCQmedium

You are troubleshooting intermittent audio quality. You notice that RTCP packets are arriving with high variance. What does this indicate?

A.Latency
B.Codec Mismatch
C.Packet Loss
D.Jitter
AnswerD

RTCP receiver reports carry interarrival jitter statistics; high variance in packet arrival times relative to the RTP timestamp indicates jitter, the variation in packet spacing. This is distinct from loss or latency, and it explains the intermittent audio degradation being troubleshot.

Why this answer

RTCP (RTP Control Protocol) provides reception quality feedback, including interarrival jitter, which is the statistical variance in packet arrival times. High variance in RTCP-reported jitter indicates network jitter — packets are arriving with uneven spacing, causing audio quality issues like choppy or robotic voice. Jitter is distinct from latency (delay) and packet loss (missing packets).

Exam trap

The trap is confusing jitter with latency or packet loss — candidates see 'high variance' and think 'delay' (latency) or 'missing packets' (loss), but RTCP jitter specifically measures interarrival variance, which is the definition of jitter.

How to eliminate wrong answers

Option A is wrong because latency is the absolute delay in packet delivery, not the variance — RTCP reports jitter as a variance metric, not a delay metric. Option B is wrong because codec mismatch would cause negotiation failures or garbled audio, not high RTCP jitter variance; RTCP jitter is a network-layer metric independent of codec. Option C is wrong because packet loss is reported separately in RTCP (fraction lost, cumulative lost) and would not manifest as high jitter variance — loss and jitter are distinct RTCP metrics.

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3
MCQmedium

When a SIP phone initiates a call, which message does the CUCM send to acknowledge the initial INVITE?

A.401 Unauthorized
B.100 Trying
C.180 Ringing
D.200 OK
AnswerB

CUCM replies with 100 Trying to the initial INVITE, a provisional response confirming the request was received and is being processed, which stops the phone retransmitting the INVITE. It precedes any 180 Ringing or 200 OK, so it is the acknowledgement the SIP phone expects first.

Why this answer

When a SIP phone sends an INVITE to CUCM, CUCM immediately returns a 100 Trying provisional response to stop the phone from retransmitting the INVITE while CUCM processes the request. This is defined in RFC 3261 as a provisional (1xx) response indicating the request has been received and is being processed. It is not a final response and does not indicate the call is ringing or answered.

Exam trap

CLCOR often tests SIP response-code sequencing, and candidates confuse the immediate 100 Trying acknowledgement with later provisional responses like 180 Ringing or final responses like 200 OK.

How to eliminate wrong answers

Option A is wrong because 401 Unauthorized is a final 4xx response used for SIP digest authentication challenges, not the immediate acknowledgement of an INVITE. Option C is wrong because 180 Ringing is a provisional response sent later, once the called party's device is actually alerting — it is not the first acknowledgement. Option D is wrong because 200 OK is the final success response sent when the call is answered, which occurs after ringing and is not an acknowledgement of the initial INVITE.

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4
MCQeasy

A network engineer is troubleshooting a Cisco IP Phone 7841 that fails to register with CUCM. The phone displays 'Configuring IP' and then 'Registering'. The engineer verifies that the phone has obtained an IP address and can ping the TFTP server. Which protocol does the phone use to download its configuration file from the TFTP server?

A.FTP
B.HTTPS
C.HTTP
D.TFTP
AnswerD

Cisco IP Phones use TFTP to download their configuration files from the CUCM TFTP server. After obtaining an IP address and discovering the TFTP server via DHCP Option 150, the phone requests its configuration file (e.g., SEP<MAC>.cnf.xml) using TFTP. This file contains the CUCM server addresses and other settings needed for registration. Without successful TFTP download, the phone cannot proceed to register.

Why this answer

Cisco IP Phones use TFTP to download their configuration files from the CUCM TFTP server. This process occurs after the phone obtains an IP address and discovers the TFTP server via DHCP Option 150. The configuration file contains the necessary information for the phone to register with CUCM, including the CUCM server list.

TFTP is the standard protocol for this step.

Exam trap

The trap here is confusing TFTP with HTTP or HTTPS, which are used for other phone functions like web access or firmware updates, but not for initial configuration download.

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5
MCQmedium

You are troubleshooting a scenario where G.711 is being used instead of the expected Opus codec between two endpoints in the same region. What is the most likely cause in the CUCM configuration?

A.Audio Codec Preference List
B.Phone NTP settings
C.SIP Trunk Security Profile mismatch
D.Location-based bandwidth limit
AnswerA

The Audio Codec Preference List defines the ordered codecs an endpoint offers during negotiation. If G.711 ranks above Opus on that list, the endpoints select G.711 despite Opus being available, directly explaining the unexpected codec in the same region.

Why this answer

The Audio Codec Preference List in CUCM defines the ordered list of codecs an endpoint will offer and accept during SIP/SDP negotiation. If Opus is not placed above G.711 (or is missing entirely) in the assigned codec preference list, the endpoints will fall back to G.711 even when both support Opus. Since both endpoints are in the same region, no location-based bandwidth constraint would force G.711, so the codec preference list is the most likely culprit.

Exam trap

CLCOR often tests the misconception that codec selection is driven by bandwidth or security settings, when in fact the Audio Codec Preference List and region configuration are the authoritative controls for codec negotiation between endpoints.

How to eliminate wrong answers

Option B is wrong because Phone NTP settings only control time synchronization for IP phones and have no bearing on which audio codec is negotiated in the SDP offer/answer. Option C is wrong because a SIP Trunk Security Profile mismatch affects TLS/SIP signaling authentication and encryption, not the media codec chosen between two registered endpoints. Option D is wrong because Location-based bandwidth limits only restrict the maximum bitrate allowed between locations; within the same region, Opus and G.711 both fit comfortably, and bandwidth limits do not reorder codec preference.

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6
MCQmedium

You are troubleshooting a registration issue where the phone sends a REGISTER request, but the server does not respond. What is the most likely issue?

A.Incorrect Model
B.Device Pool mismatch
C.Wrong Password
D.Network Connectivity/Firewall
AnswerD

A REGISTER request that reaches the server but draws no response indicates the reply is blocked or the path is broken. Firewall rules or routing commonly drop SIP signalling, satisfying the stem's no-response symptom, whereas authentication failures would return a 4xx rejection.

Why this answer

If the phone sends a REGISTER request but the server does not respond at all, the most likely cause is a network connectivity or firewall issue blocking the traffic. If the server received the request, it would typically respond with a 401 Unauthorized challenge or a 200 OK. A complete lack of response indicates the packet is not reaching the server or the response is blocked.

Exam trap

CLCOR often tests the difference between no response and an error response; candidates pick wrong password because they assume authentication failure, but a wrong password yields a 401/403, not silence.

How to eliminate wrong answers

Option A is wrong because an incorrect phone model would not prevent the server from responding; the phone would still send a REGISTER and the server would process it, possibly rejecting with an error. Option B is wrong because a device pool mismatch is a configuration issue that would typically result in the server responding with an error or the phone failing to register after a response, not a total lack of response. Option C is wrong because a wrong password would cause the server to respond with a 401 Unauthorized or 403 Forbidden, not silence.

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7
Multi-Selecteasy

Which TWO features are provided by the Cisco Webex App when registered as a Softphone?

Select 2 answers
A.TFTP service hosting
B.SIP signaling
C.DNS root server
D.RTP media transmission
E.DHCP server functionality
AnswersB, D

Required for call control.

Why this answer

The Cisco Webex App, when registered as a softphone, functions as a SIP endpoint, so it performs SIP signaling (Option B) to set up, manage, and tear down calls with the call control platform. It also handles RTP media transmission (Option D), sending and receiving the actual audio/video packets for the call. These two functions are core to any softphone client, which must both signal and carry media.

Options A, C, and E are incorrect because a softphone does not host TFTP configuration files, act as a DNS root server, or provide DHCP address assignment; those are infrastructure services typically provided by dedicated servers or network devices.

Exam trap

CLCOR often tests the distinction between endpoint functions (SIP signaling, RTP media) and network infrastructure services (TFTP, DNS, DHCP), so candidates may mistakenly select infrastructure roles that a softphone does not provide.

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8
MCQhard

An endpoint sends a SIP REGISTER request. CUCM returns a 401 Unauthorized response. What is the next logical step in the SIP signaling flow for the endpoint?

A.Switch to SCCP
B.Resend REGISTER with Authorization header
C.Terminate the TCP connection
D.Send a SIP INVITE
AnswerB

A 401 Unauthorized response is a challenge demanding credentials, so the endpoint must resend REGISTER including an Authorization header carrying its digest credentials. This directly satisfies the challenge-response authentication constraint in the stem, completing the SIP registration handshake rather than retrying blindly or abandoning the attempt.

Why this answer

When CUCM returns 401 Unauthorized to a SIP REGISTER, it is issuing a challenge under SIP digest authentication (RFC 3261). The endpoint must compute the digest response using its credentials and the challenge nonce, then resend the REGISTER with an Authorization header containing that response. This is the standard challenge-response handshake for SIP registration.

Exam trap

CLCOR often tests whether candidates understand that a 401 is a challenge, not a failure — the trap is treating it as a terminal error and choosing to terminate the connection or switch protocols instead of resending with credentials.

How to eliminate wrong answers

Option A is wrong because switching to SCCP is not a SIP signaling step — SCCP is a different protocol entirely, and endpoints do not switch protocols mid-registration in response to a 401. Option C is wrong because terminating the TCP connection would abort the registration attempt; the 401 is a challenge, not a fatal error, so the connection should remain open for the authenticated retry. Option D is wrong because a SIP INVITE is used to initiate a session (call), not to complete registration; sending an INVITE before successful registration would fail because the endpoint is not yet registered.

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9
MCQmedium

A Cisco IP Phone 8845 in the New York office registers to CUCM and downloads its configuration. The administrator notices that the phone uses a DHCP scope option to discover the TFTP server address. Which DHCP option is used by the phone for this purpose?

A.Option 42
B.Option 150
C.Option 43
D.Option 66
AnswerB

DHCP Option 150 provides a list of TFTP server IP addresses to Cisco IP phones during network boot. In this scenario, the phone uses Option 150 to locate the TFTP server from which it downloads its configuration and firmware. This is the standard method for Cisco endpoints to discover the TFTP service in a CUCM environment, allowing the phone to proceed with registration.

Why this answer

Cisco IP phones use DHCP Option 150 to obtain a list of TFTP server addresses. This allows the phone to contact the correct TFTP server to download its configuration file and firmware. In a CUCM environment, Option 150 is the standard method for phone boot and registration, ensuring the device can locate the necessary files to become operational.

Exam trap

The trap here is confusing DHCP Option 150 with Option 66, as both can provide TFTP server information, but Cisco phones specifically use Option 150.

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10
MCQeasy

When configuring an IP phone for automatic provisioning, which DHCP option is most commonly used to provide the TFTP server address to the phone?

A.Option 150
B.Option 43
C.Option 66
D.Option 12
AnswerA

Option 150 returns the TFTP server address as a single IP in the vendor-specific field, which Cisco IP phones read directly for provisioning. Option 66 carries it as a hostname string, which phones do not reliably parse.

Why this answer

DHCP Option 150 is the Cisco-proprietary option most commonly used to deliver a list of TFTP server IP addresses to Cisco IP phones during auto-provisioning. The phone uses this TFTP address to download its configuration file (SEP<MAC>.cnf.xml) and firmware.

Exam trap

CLCOR often tests the confusion between Option 150 (Cisco TFTP IP list) and Option 66 (generic single TFTP server name), causing candidates to pick the more 'standard' option.

How to eliminate wrong answers

Option B is wrong because Option 43 is used for vendor-specific information (commonly for wireless access point controllers and some vendor encapsulation), not the standard Cisco phone TFTP delivery. Option C is wrong because Option 66 provides a single TFTP server name (often a hostname/FQDN) and is more common in non-Cisco VoIP environments; Cisco phones prefer Option 150's IP list. Option D is wrong because Option 12 supplies the client hostname, not a TFTP server address.

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11
MCQmedium

A Cisco IP Phone 8845 registered to CUCM is not receiving its configuration file from the TFTP server after a factory reset. The phone has received its IP address via DHCP and can ping the TFTP server. Which option in the DHCP scope must be configured to allow the phone to locate the TFTP server?

A.Option 42
B.Option 66
C.Option 3
D.Option 150
AnswerD

Option 150 provides the TFTP server address to Cisco IP Phones. Without it, the phone cannot locate the TFTP server to download its configuration file. This is the correct answer because the phone has network connectivity but lacks the TFTP server information, which is typically delivered via DHCP option 150 in Cisco environments.

Why this answer

Cisco IP Phones rely on DHCP option 150 to learn the IP address of the TFTP server that hosts their configuration files. Without this option, the phone cannot proceed with provisioning. The other options serve different purposes: option 66 is a non-standard alternative, option 42 is for NTP, and option 3 is for the default gateway.

Only option 150 directly addresses the phone's need to locate the TFTP server.

Exam trap

The trap here is confusing DHCP option 150 with option 66, as both can provide TFTP server information, but Cisco IP Phones specifically require option 150.

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12
MCQmedium

A collaboration engineer is troubleshooting a one-way audio issue between two Cisco IP Phones registered to the same Cisco Unified Communications Manager cluster. The phones are in different locations separated by a WAN link. The engineer captures traffic and notices that the SIP signaling is correct, but RTP packets are only flowing in one direction. The engineer suspects a NAT or firewall issue. Which protocol or feature should the engineer verify to ensure that RTP can traverse the NAT and firewalls correctly?

A.SIP Early Offer
B.Media Termination Point (MTP)
C.Skinny Client Control Protocol (SCCP)
D.Interactive Connectivity Establishment (ICE)
AnswerD

ICE is a protocol that enables NAT traversal for media streams by discovering the best path between endpoints. It uses STUN and TURN servers to gather candidate addresses and perform connectivity checks. In a CUCM environment, ICE can be enabled on SIP profiles to allow endpoints to negotiate media through NAT. For one-way audio due to NAT, verifying ICE configuration is essential. This is the correct feature to ensure RTP traverses NAT.

Why this answer

One-way audio often results from NAT or firewall blocking RTP in one direction. ICE is a protocol that helps endpoints discover and negotiate media paths through NAT by using STUN and TURN. In CUCM, ICE can be enabled on SIP profiles to facilitate NAT traversal.

Verifying ICE configuration is the correct step to ensure RTP flows bidirectionally. Other options like Early Offer or MTP do not directly address NAT traversal for media.

Exam trap

The trap here is assuming that any media-related feature like MTP or Early Offer will solve NAT traversal, when ICE is specifically designed for that purpose.

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13
MCQeasy

Which protocol is primarily used for the transmission of real-time voice and video data between two endpoints?

A.SCCP
B.TFTP
C.RTP
D.SIP
AnswerC

RTP carries real-time voice and video media between endpoints, running over UDP to avoid retransmission delays. It provides the sequencing, timestamps and payload type identification needed for streaming audio and video, while signalling protocols such as SIP establish the session.

Why this answer

RTP (Real-time Transport Protocol) is the protocol that actually carries the digitized voice and video media streams between endpoints in real-time communications. It runs over UDP to minimize latency and jitter, and is paired with RTCP for quality feedback. SIP, SCCP, and TFTP are signaling or provisioning protocols, not media transport.

Exam trap

CLCOR often tests the distinction between signaling protocols (SIP, SCCP) and media transport protocols (RTP), so candidates who see 'voice and video' and think 'SIP' pick the wrong answer.

How to eliminate wrong answers

Option A is wrong because SCCP (Skinny Client Control Protocol) is a Cisco proprietary signaling protocol used between Cisco IP phones and CUCM for call setup and control, not for carrying media. Option B is wrong because TFTP is a trivial file transfer protocol used to download configuration files and firmware to IP phones, not to transmit real-time voice/video. Option D is wrong because SIP (Session Initiation Protocol) is a signaling protocol for establishing, modifying, and tearing down sessions — it negotiates the media but does not transport it.

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14
MCQeasy

A network administrator is configuring a Cisco IP Phone 7811 to register with Cisco Unified Communications Manager. The phone is powered on and connected to the network. The administrator notices that the phone is not obtaining an IP address. Which protocol is responsible for providing the phone with an IP address and other network parameters?

A.TFTP
B.DHCP
C.SIP
D.CDP
AnswerB

DHCP (Dynamic Host Configuration Protocol) is used by Cisco IP Phones to obtain an IP address, subnet mask, default gateway, and TFTP server address. Without DHCP, the phone cannot communicate on the network. In this scenario, the phone is not obtaining an IP address, so DHCP is the protocol that should be providing it. This is a fundamental step in the phone boot process.

Why this answer

Cisco IP Phones use DHCP to obtain an IP address and other network settings such as the TFTP server address and default gateway. Without DHCP, the phone cannot proceed with the boot process. TFTP is used later to download configuration files, CDP is for neighbor discovery, and SIP is for call signaling.

Therefore, DHCP is the correct protocol.

Exam trap

The trap here is confusing the order of the boot process, thinking that TFTP or CDP might provide IP addressing when they serve different purposes.

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15
Multi-Selecthard

A collaboration engineer is deploying a new Cisco Webex Room Kit Pro in a conference room. The device is registered to Cisco Unified Communications Manager (CUCM) using SIP. The engineer needs to ensure that the device can place calls to external numbers and that the dialed digits are correctly processed. Which two configuration elements must be defined on CUCM to enable external dialing from this endpoint? (Choose two.)

Select 2 answers
A.SIP Profile with Outbound Proxy
B.Route Pattern
C.Device Pool with Region settings
D.Gateway or Route List
E.Translation Pattern
AnswersB, D

A Route Pattern in CUCM defines how dialed digits are routed to external destinations. It matches the dialed string and points to a Route List or Gateway. Without a Route Pattern, the call cannot be routed outside the cluster. For external dialing, the Route Pattern must be configured with the appropriate dial plan, such as 9.@ for outside access. This is essential for the Room Kit Pro to reach external numbers.

Why this answer

To enable external dialing from a Cisco Webex Room Kit Pro registered to CUCM, you must configure a Route Pattern to match the dialed digits and route the call, and a Gateway or Route List to provide the physical connection to the external network. The Route Pattern directs the call to the Route List, which then selects a Gateway. Without these, the call cannot leave the CUCM cluster.

Exam trap

The trap here is confusing dial plan manipulation elements like Translation Patterns with the core routing elements required for external dialing.

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16
MCQeasy

Which protocol provides the signaling for Webex Room devices when they are registered to CUCM (on-premises)?

A.MGCP
B.SCCP
C.SIP
D.H.323
AnswerC

SIP carries the call signalling — registration, invite and session setup — between Webex Room devices and CUCM. Registering on-premises requires this signalling channel, so SIP satisfies the stem's requirement, while media itself flows separately over RTP.

Why this answer

Webex Room devices, when registered on-premises to CUCM, use SIP (Session Initiation Protocol) for signaling. SIP is the standard protocol for establishing, modifying, and terminating multimedia sessions, and it is supported by Webex devices for call control with CUCM.

Exam trap

CLOOR often tests the confusion between signaling protocols, where candidates might think SCCP or MGCP is used for Webex devices, but SIP is the correct protocol.

How to eliminate wrong answers

Option A is wrong because MGCP (Media Gateway Control Protocol) is used for gateway control, not for endpoint signaling by Webex devices. Option B is wrong because SCCP (Skinny Client Control Protocol) is a Cisco proprietary protocol used by older IP phones, but Webex Room devices do not support SCCP. Option D is wrong because H.323 is an older suite of protocols for multimedia communication, but Webex devices use SIP for CUCM registration.

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17
MCQmedium

A Cisco Webex Room Kit is registered to CUCM on-premises. The administrator wants to enable encryption for signaling and media. Which protocol does the device use for secure signaling when registered to CUCM?

A.HTTPS
B.SIP over TLS
C.TLS for media
D.SRTP
AnswerB

SIP over TLS is used for secure signaling between Cisco Webex Room devices and CUCM. When encryption is enabled, the device uses TLS to encrypt SIP messages, ensuring that signaling information such as call setup and teardown is protected. This is part of a secure collaboration environment and is supported by CUCM for endpoints that support TLS, including Webex Room devices.

Why this answer

Cisco Webex Room devices use SIP over TLS for secure signaling when registered to CUCM. This encrypts the SIP messages, protecting call control information. For media encryption, SRTP is used.

Together, they provide a secure collaboration environment. The administrator should ensure that CUCM is configured with a secure SIP profile and that the device trusts the CUCM certificate.

Exam trap

The trap here is confusing signaling encryption with media encryption; SRTP is for media, while SIP over TLS is for signaling.

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18
MCQeasy

Which component in the Cisco Collaboration architecture provides the initial configuration files (XML) to an IP phone upon registration?

A.CallManager Service
B.TFTP Service
C.CTI Manager
D.Webex Control Hub
AnswerB

The TFTP service delivers the XML configuration files an IP phone requests during boot, satisfying the stem's requirement for initial provisioning on registration. Phones retrieve their SEP<MAC>.cnf.xml and dial-plan definitions from this service before registering with Cisco Unified Communications Manager, making it the correct component.

Why this answer

The Cisco TFTP service, running on Cisco Unified Communications Manager, is responsible for serving XML configuration files, phone firmware, and other provisioning files to IP phones during registration. When a phone boots, it requests its SEP<MAC>.cnf.xml file from the TFTP server, which contains its directory number, device pool, and cluster settings. Without TFTP, phones cannot complete registration.

Exam trap

CLCOR often tests whether candidates confuse the CallManager service (call control) with the TFTP service (provisioning) — registration signaling and configuration file delivery are handled by different services.

How to eliminate wrong answers

Option A is wrong because the CallManager service (Cisco CallManager) handles call signaling, registration, and call control — it does not serve configuration files. Option C is wrong because CTI Manager provides JTAPI/TAPI interfaces for applications like contact centers and does not handle phone provisioning. Option D is wrong because Webex Control Hub is the cloud management portal for Webex, not the on-premises TFTP provisioning service for CUCM-registered phones.

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19
MCQmedium

You are monitoring RTP traffic using a protocol analyzer. You observe that the Payload Type field in the RTP header is set to 0. Which codec is currently in use?

A.G.729
B.Opus
C.G.711 PCMU
D.G.711 PCMA
AnswerC

Payload Type 0 is statically mapped to G.711 µ-law (PCMU) in the RTP audio/video profile, so the analyser reading confirms PCMU is in use. The PT field is the header's codec identifier, and value 0 is reserved for this 64 kbps narrowband codec.

Why this answer

In the RTP header, the Payload Type (PT) field value of 0 indicates the use of the G.711 PCMU (μ-law) codec. This is defined in RFC 3551, which assigns static payload type 0 to PCMU. Therefore, the correct answer is G.711 PCMU.

Exam trap

CLCOR often tests the static payload type assignments, and candidates may confuse PT 0 (PCMU) with PT 8 (PCMA) or assume dynamic types for all codecs.

How to eliminate wrong answers

Option A is wrong because G.729 is assigned a dynamic payload type (typically 18) and is not represented by PT 0. Option B is wrong because Opus uses dynamic payload types (e.g., 96-127) and is not assigned a static PT. Option D is wrong because G.711 PCMA (A-law) is assigned static payload type 8, not 0.

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20
Multi-Selectmedium

Which THREE network conditions contribute to poor VoIP call quality in an RTP stream?

Select 3 answers
A.Latency
B.High MTU
C.Jitter
D.Packet Loss
E.Fast switching
AnswersA, C, D

Delay causes conversation overlaps.

Why this answer

Latency (A) is correct because one-way delay above roughly 150 ms degrades RTP voice interactivity, causing talk-over and echo that the jitter buffer cannot fix. Jitter (C) is correct because variable inter-packet arrival times force the jitter buffer to either underrun (clipping) or grow (added delay), directly harming voice quality. Packet Loss (D) is correct because dropped RTP packets cannot be retransmitted in real time, so codecs conceal the gaps, producing robotic or choppy audio.

High MTU (B) is not inherently harmful—fragmentation is the real risk—and Fast switching (E) is a LAN forwarding behavior, not a WAN impairment affecting RTP quality.

Exam trap

CLCOR often tests whether candidates confuse general network performance factors (MTU, switching mode) with the specific RTP impairments defined by RFC 3550 — only latency, jitter, and packet loss belong in that list.

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