CLCOR Protocols And Endpoints Practice Test 3 — 15 Questions

Question 1

A network engineer is configuring a new Cisco IP Phone 8845 to register with CUCM. The phone is on a VLAN that has DHCP configured with Option 150 pointing to the CUCM TFTP server. After the phone boots, it displays 'Configuring CM List' and then 'Registering'. However, the phone remains in this state and never completes registration. The engineer verifies that the phone can ping the TFTP server and that the TFTP server is reachable. Which action should the engineer take next to troubleshoot the registration failure?

Choose an answer to begin — your selection is scored in the full session.

Accepted Answer

Verify that the phone's MAC address is configured in CUCM and that the device is associated with a valid directory number.. When a Cisco IP Phone displays 'Registering' but does not complete, the most common cause is that the phone's MAC address is not configured in CUCM or the device is not associated with a directory number. The phone has already obtained network settings and downloaded its configuration, so the next step is to verify the device configuration in CUCM. This ensures the phone is authorized to register.

Answer

Check the DHCP scope to ensure Option 150 is correctly pointing to the TFTP server's IP address.

Answer

Verify that the phone's firmware is compatible with the CUCM version by checking the Device Defaults configuration.

Answer

Restart the TFTP service on the CUCM server to ensure the configuration files are available.

Question 2

A collaboration engineer is analyzing a SIP call flow between two CUCM clusters. The engineer observes that the SIP INVITE from the originating cluster includes a Diversion header with the original called number, but the terminating cluster does not use this information to route the call correctly. The engineer wants to ensure that the terminating cluster uses the Diversion header for call routing. Which configuration should the engineer implement on the terminating cluster?

Accepted Answer

Configure the SIP trunk on the terminating cluster to use the Diversion header for inbound calls by enabling 'Use Diverting Party for Call Routing' in the trunk's Inbound Calls settings.. To make CUCM use the Diversion header for call routing, you must enable the 'Use Diverting Party for Call Routing' option in the SIP trunk's Inbound Calls configuration. This setting tells CUCM to extract the diverting number from the Diversion header and use it as the called number for routing decisions. This is the standard method for handling redirected calls across clusters.

Answer

Modify the route pattern on the terminating cluster to match the Diversion header number and route the call.

Answer

Enable 'Use Diverting Party for Call Routing' in the SIP Profile on the terminating cluster.

Answer

Create a SIP normalization script to copy the Diversion header value into the From header before routing.

Question 3

A collaboration engineer is configuring a new SIP trunk between CUCM and a Cisco Expressway-C for Mobile and Remote Access (MRA). The engineer needs to ensure that the SIP trunk supports secure signaling and media, and that it can handle calls to remote endpoints. Which two configurations are required on the SIP trunk to enable secure signaling and media? (Choose two.)

Accepted Answer

Set the SIP trunk security profile to use TLS for signaling and SRTP for media.. To enable secure signaling and media on a SIP trunk to Expressway-C, you must configure the SIP trunk security profile to use TLS for signaling and SRTP for media, and ensure a valid X.509 certificate is installed for TLS authentication. These two settings provide encryption and authentication for the trunk, which is essential for MRA calls.

Answer

Assign a Media Termination Point (MTP) to the SIP trunk to handle SRTP encryption.

Answer

Enable 'Allow Incoming SIP Calls' on the SIP trunk security profile.

Answer

Configure the SIP trunk to use a secure SIP URL (sips) for the destination address.

Question 4

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?

Accepted Answer

Resend REGISTER with Authorization header. 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.

Answer

Switch to SCCP

Answer

Terminate the TCP connection

Answer

Send a SIP INVITE

Question 5

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

Accepted Answer

Option 150. 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.

Answer

Option 43

Answer

Option 66

Answer

Option 12

Question 6

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?

Accepted Answer

Audio Codec Preference List. 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.

Answer

Phone NTP settings

Answer

SIP Trunk Security Profile mismatch

Answer

Location-based bandwidth limit

Question 7

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

Accepted Answer

RTP. 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.

Answer

SCCP

Answer

TFTP

Answer

SIP

Question 8

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?

Accepted Answer

G.711 PCMU. 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.

Answer

G.729

Answer

Opus

Answer

G.711 PCMA

Question 9

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

Accepted Answer

Jitter. 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).

Answer

Latency

Answer

Codec Mismatch

Answer

Packet Loss

Question 10

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

Accepted Answer

TFTP Service. 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.

Answer

CallManager Service

Answer

CTI Manager

Answer

Webex Control Hub

Question 11

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

Accepted Answer

100 Trying. 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.

Answer

401 Unauthorized

Answer

180 Ringing

Answer

200 OK

Question 12

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

Accepted Answer

SIP TLS 5061. 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.

Answer

HTTP 443

Answer

SIP TCP 5060

Answer

SRTP 5061

Question 13

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

Accepted Answer

SIP. 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.

Answer

MGCP

Answer

SCCP

Answer

H.323

Question 14

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?

Accepted Answer

Network Connectivity/Firewall. 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.

Answer

Incorrect Model

Answer

Device Pool mismatch

Answer

Wrong Password

Question 15

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

Accepted Answer

SIP signaling. 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.

Answer

TFTP service hosting

Answer

DNS root server

Answer

DHCP server functionality