CLCOR Protocols And Endpoints — Timed 30 Questions

Question 1

A Webex Desk Pro is failing to provision via the Cisco Webex Control Hub. The device displays 'Provisioning Error'. Which configuration component is primarily responsible for holding the authentication credentials required for the device to pull its configuration?

30 minute time limit — choose an answer to begin.

Accepted Answer

Activation Code. The Activation Code is the primary component that holds the authentication credentials for a Webex Desk Pro to provision via Control Hub. When entered on the device, it securely associates the device with the organization and allows it to pull its configuration. Without a valid activation code, the device cannot authenticate to Control Hub.

Answer

TFTP Address

Answer

SIP Digest User

Answer

VLAN ID

Question 2

An endpoint using the Opus codec is experiencing one-way audio. The SDP exchange indicates that the endpoint is offering a wideband range. Where in CUCM would you modify the Region settings to force the use of a specific codec for this endpoint?

Accepted Answer

Region Settings. Codec selection between regions is controlled by CUCM Region settings, where you configure the maximum audio codec (e.g., G.722, Opus, G.711) allowed for calls between devices in those regions. To force a specific codec for the endpoint, you modify the Region configuration that applies to its device pool's region pair.

Answer

Service Parameters

Answer

Device Pool

Answer

SIP Profile

Question 3

A Cisco IP Phone 8841 is failing to register to Cisco Unified Communications Manager. You check the phone settings and see the TFTP server address is correct. Which protocol and port should you monitor on the gateway or firewall to verify the initial registration attempt?

Accepted Answer

SIP UDP 5060. The Cisco IP Phone 8841 is a SIP endpoint by default, and its initial registration signaling to Cisco Unified Communications Manager uses SIP over UDP port 5060 (or TCP 5060). Since the TFTP address is already correct, the next step in the registration sequence is the SIP REGISTER exchange, so monitoring SIP UDP 5060 on the firewall/gateway verifies whether the registration attempt reaches CUCM.

Answer

SCCP 2000

Answer

TFTP 69

Answer

HTTP 8080

Question 4

You are configuring a G.729 codec for a WAN link. To ensure that the voice quality remains acceptable while minimizing bandwidth, which feature must be enabled to prevent excessive packet overhead?

Accepted Answer

Payload size adjustment. VAD (Voice Activity Detection) suppresses silence, saving bandwidth, but is often recommended to be disabled if voice quality is the primary concern, or enabled to conserve bandwidth.

Answer

DSCP marking

Answer

Jitter buffer sizing

Answer

RTCP feedback

Question 5

You need to verify that an endpoint is correctly sending RTCP reports to Cisco Unified Communications Manager. Which field in the RTCP packet header would you examine to track the jitter statistics?

Accepted Answer

Interarrival Jitter. The Interarrival Jitter field in the RTCP packet header carries the jitter statistics, which measure the variation in packet arrival times. This field is part of the RTCP Receiver Report (RR) and is used to assess media stream quality. Examining it allows you to verify that endpoints are reporting jitter correctly to CUCM.

Answer

Sequence Number

Answer

SSRC

Answer

Timestamp

Question 6

A SIP endpoint is behind a NAT device. The endpoint registers successfully, but audio is one-way. What feature should be enabled on the SIP Profile to handle this issue?

Accepted Answer

Media Termination Point (MTP) Required. The 'SIP Rel1XX' or 'SIP OPTIONS Ping' are common, but 'NAT Support' or 'SIP Privacy' settings are often required. Specifically, 'SIP Profile' configuration to support NAT traversal (media termination point usage) is key.

Answer

SIP OPTIONS Ping

Answer

Disable DTMF Relay

Answer

Enable Early Media

Question 7

You need to support Opus codec on a Cisco IP Phone 8865. Which configuration object must be updated to allow this codec?

Accepted Answer

Audio Codec Preference List. Opus is a codec, and in Cisco Unified Communications Manager codec support is governed by the Audio Codec Preference List, which defines which codecs a device offers and in what priority. To allow Opus on an 8865, you must add Opus to the codec preference list assigned to the device or device pool. The SIP Profile, Common Phone Profile, and Device Defaults do not control codec availability.

Answer

SIP Profile

Answer

Common Phone Profile

Answer

Device Defaults

Question 8

Which field in the SIP INVITE SDP body specifies the media attributes such as codec type?

Accepted Answer

m= line. The 'm=' line in the SDP body of a SIP INVITE specifies the media type, port, and protocol, and also includes media attributes such as codec type (e.g., audio codec payload types). It is the primary field that describes the media stream and its characteristics.

Answer

o= line

Answer

c= line

Answer

s= line

Question 9

An IP phone is stuck in a boot loop. You suspect a corrupted firmware image. Which process would you initiate to force the phone to download a new image?

Accepted Answer

Factory Reset. A factory reset is the process that forces an IP phone to download a new firmware image from the TFTP server. When a phone is stuck in a boot loop due to corrupted firmware, a factory reset clears the current configuration and triggers a fresh firmware download. Therefore, 'Factory Reset' is the correct answer.

Answer

Restart Device

Answer

Refresh CTI

Answer

Reset SIP Profile

Question 10

Which TWO protocols are used for managing the quality and monitoring of VoIP sessions?

Accepted Answer

RTCP. RTCP (RTP Control Protocol) is correct because it works alongside RTP to provide out-of-band statistics and control information for an RTP session, delivering quality-of-service feedback such as packet loss, jitter, and round-trip delay used to monitor and manage VoIP quality. SNMP (Simple Network Management Protocol) is correct because it is the standard framework for monitoring and managing network devices and services, including polling VoIP gateways, IP phones, and call managers for performance and fault data. TFTP is a simple file-transfer protocol used for things like provisioning configs and firmware, not for session quality monitoring. SIP is a signaling protocol for establishing, modifying, and terminating VoIP sessions, not for quality monitoring. RTP carries the actual media streams but does not itself provide the quality statistics and management feedback that RTCP and SNMP do.

Answer

TFTP

Answer

SIP

Answer

RTP

Question 11

Which THREE codecs are natively supported by the Cisco IP Phone 8800 series?

Accepted Answer

Opus. The Cisco IP Phone 8800 series natively supports Opus (A), a wideband codec used for high-quality audio in Webex and SIP environments, G.729 (C), a low-bandwidth compressed codec commonly used for WAN-efficient voice calls, and G.711 (E), the standard 64 kbps PCM codec (a-law/mu-law) that provides toll-quality audio and is universally supported. These three are part of the phone's built-in codec set for SIP and Webex calling. H.264 (B) is a video codec, not an audio codec, and while the 8800 series supports video, the question asks about natively supported codecs in the voice codec list; it is not among the three marked correct. G.723 (D) is an older low-bit-rate codec that is not natively supported on the 8800 series.

Answer

H.264

Answer

G.723

Question 12

To ensure that G.729 is used for all inter-site calls but G.711 is used for intra-site calls, what specific CUCM construct must you define?

Accepted Answer

Region Relationship. A Region Relationship in Cisco Unified Communications Manager defines the codec used between two regions. To force G.729 between sites and G.711 within a site, you assign each site to a region and configure the region-pair relationship to use G.729 for inter-region calls while the intra-region (same region) setting uses G.711. This is the specific construct that controls codec selection based on call path.

Answer

Location

Answer

Device Pool

Answer

Transcoder

Question 13

Which TWO tasks must be performed when moving a phone from one CUCM cluster to another?

Accepted Answer

Reset the phone. Option A (Reset the phone) is correct because after the phone's configuration is changed to point to the new cluster, the device must be reset so it re-registers and downloads its new configuration from the new CUCM. Option C (Change TFTP Server address) is correct because the phone must be given the IP address or hostname of the new cluster's TFTP server so it can download the correct configuration file and firmware. Option B (Enable SIP OPTIONS) is not required for the move itself; SIP OPTIONS is a monitoring/keepalive mechanism, not a migration step. Option D (Update the CTI Manager) is not a phone migration task; CTI Manager is a CUCM service used by applications, not something reconfigured on the phone. Option E (Delete the phone from AD) is irrelevant because CUCM phone records are stored in the CUCM database, not Active Directory.

Answer

Enable SIP OPTIONS

Answer

Update the CTI Manager

Answer

Delete the phone from AD

Question 14

Which THREE methods are used by Cisco IP phones to obtain their configuration files?

Accepted Answer

Auto-discovery via Service Discovery. Cisco IP phones can locate their TFTP configuration server through Service Discovery, which is the auto-discovery mechanism (Option C) that lets the phone learn the TFTP server address automatically. Option D, Manual TFTP entry, is correct because a phone can be manually configured with the TFTP server IP address, for example by pressing Settings and entering the TFTP server address directly on the phone. Option E, DHCP Option 150, is correct because this DHCP option supplies the phone with the TFTP server IP address (or a list of addresses) during IP address assignment, which is the most common deployment method. SMTP (Option A) is a mail transfer protocol and has nothing to do with phone configuration retrieval, and FTP (Option B) is a general file transfer protocol that Cisco IP phones do not use to download their configuration files; they use TFTP instead.

Answer

SMTP

Answer

FTP

Question 15

Which THREE items are included in a standard SIP REGISTER request?

Accepted Answer

To Header. The To header (B) is mandatory in a SIP REGISTER request because it carries the address-of-record (AOR) whose bindings are being registered, e.g. To: <sip:alice@example.com>. The From header (D) is also required and identifies the registrant submitting the request, typically matching the AOR and containing a tag parameter for dialog identification. The Contact header (E) is essential because it supplies the binding information — the SIP URI (and optional q value and expires parameter) where the user agent can be reached, which is the core purpose of REGISTER. Alert-Info (A) is not part of a standard REGISTER; it is used in INVITE responses to specify ringback tones. A body with SDP (C) is not included in a standard REGISTER since REGISTER does not negotiate media sessions; SDP is carried in INVITE and its responses.

Answer

Alert-Info

Answer

Body (SDP)

Question 16

Which TWO parameters are defined within a SIP Profile in CUCM?

Accepted Answer

MTP required flag. SIP timers and feature flags (like MTP requirements) are defined here.

Answer

Device Pool

Answer

Location bandwidth

Answer

Codec Preference

Question 17

Which TWO factors significantly impact the choice of audio codec during a call setup in CUCM?

Accepted Answer

Codec Preference List. Option A (Codec Preference List) is correct because CUCM uses the codec preference list configured on the device pool (or the phone's own codec list) to determine which codecs are offered and in what priority order during call setup, directly shaping the codec negotiation. Option D (Region configuration) is correct because regions define the maximum audio codec bit rate (and thus which codecs are permitted) between pairs of regions, so the calling and called parties' regions determine the allowed codec set for the call. Option B (DNS Server) is not a codec-selection factor; DNS resolves hostnames for endpoints and services but does not influence media codec negotiation. Option C (VLAN ID) only affects Layer 2 network segmentation and QoS/voice VLAN placement, not which codec is chosen. Option E (NTP Server) provides time synchronization for certificates, logs, and CDR timestamps, and has no role in codec selection during call setup.

Answer

DNS Server

Answer

VLAN ID

Answer

NTP Server

Question 18

A Cisco TelePresence endpoint registered to CUCM is experiencing one-way audio. The endpoint is behind a NAT device, and the SIP signaling is working correctly. Which configuration change on CUCM is most likely to resolve the one-way audio issue?

Accepted Answer

Configure the SIP profile to allow the endpoint to send its public IP address in SDP.. One-way audio behind NAT typically happens when the endpoint includes its private IP address in the SDP, which the far end cannot reach. The SIP profile must be configured to allow the endpoint to advertise its public IP address in SDP, ensuring that media flows correctly in both directions.

Answer

Configure the SIP profile to use the 'Outbound Call Route' to the public IP.

Answer

Enable 'Media Termination Point Required' on the SIP trunk.

Answer

Enable 'SIP Normalization' on the SIP trunk.

Question 19

A Cisco Webex Room Kit is registered to CUCM using SIP. During a call to an external H.323 endpoint via a Cisco Expressway, the call connects but no video is seen. Which two actions should you take to troubleshoot the issue? (Choose two.)

Accepted Answer

Verify the SIP profile assigned to the device includes the correct early offer configuration.. When a call connects but video fails, the issue often lies in media negotiation or bandwidth restrictions. The SIP profile's early offer settings determine whether video capabilities are included in the initial SDP. Region settings define the maximum video bandwidth and codec allowed between devices. Both must be correct to establish video. The other options are less likely: the SIP port is already working, changing protocol is not a fix, and MTPs do not handle video.

Answer

Ensure the device is configured to use the H.323 protocol instead of SIP.

Answer

Confirm that the Webex Room Kit is using the default SIP port 5060.

Answer

Verify that the MTP is configured to pass video.

Question 20

An administrator is configuring a SIP trunk between CUCM and a Cisco Unified Border Element (CUBE) to support early offer for outbound calls. Which two SIP message headers are essential for the early offer mechanism to function correctly? (Choose two.)

Accepted Answer

Supported: 100rel. For early offer to work, the initial INVITE must include an SDP offer, indicated by the Content-Type: application/sdp header. Additionally, the Supported: 100rel header signals that the endpoint can handle reliable provisional responses, which are used to confirm media setup before the final answer. Together, these headers enable the early establishment of the media path.

Answer

Allow: INVITE, ACK, CANCEL, BYE

Answer

Content-Disposition: session

Answer

Require: 100rel