CLCOR Protocols And Endpoints Practice Test 2 — 15 Questions

Question 1

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 2

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 3

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 4

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 5

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

Question 6

A Cisco Webex Room Kit registered to Cisco Unified Communications Manager (CUCM) using SIP is not receiving an incoming call from a third-party SIP gateway. The CUCM traces show that CUCM sends a SIP INVITE to the endpoint with a single codec offer of G.711ulaw, but the endpoint rejects the INVITE with a 488 Not Acceptable Here response. The endpoint's SIP profile is configured with the default settings. Which action should you take to resolve this issue?

Accepted Answer

Ensure that the region assigned to the endpoint includes G.711ulaw and that the endpoint's device pool is correctly associated with that region.. The 488 Not Acceptable Here response from the endpoint indicates a codec mismatch. CUCM uses regions to determine which codecs are offered between devices. If the region assigned to the endpoint does not include G.711ulaw, CUCM will not offer it, and the endpoint will reject the call. Therefore, verifying and correcting the region configuration to include G.711ulaw and ensuring the device pool is associated with that region is the proper resolution.

Answer

Enable the 'Accept Unknown Codecs' option on the SIP profile to allow the endpoint to accept the G.711ulaw offer.

Answer

Configure a SIP normalization script on the SIP trunk to add the G.711ulaw codec to the INVITE.

Answer

Modify the SIP profile assigned to the endpoint to include the G.711ulaw codec in the Allowed Codecs list.

Question 7

A collaboration engineer is analyzing a SIP call flow between two CUCM clusters. The engineer observes that the SIP INVITE from the originating cluster contains a 'Diversion' header, but the terminating cluster does not process it. Which SIP header is used to convey the original called number when a call is forwarded?

Accepted Answer

History-Info. The History-Info header is the standard SIP header used to convey the original called number and the history of call forwarding. It allows the terminating cluster to understand the call's path and the original destination. CUCM supports History-Info for intercluster calls, enabling features like diverting and forwarding to work correctly across clusters.

Answer

Remote-Party-ID

Answer

Diversion

Answer

P-Asserted-Identity

Question 8

A Cisco IP Phone 8845 is registered to CUCM and is configured to use the Opus codec for calls. During a call with a Cisco Webex Room Kit, the phone sends an RTP stream with payload type 114, but the Room Kit responds with a 488 Not Acceptable Here. The CUCM traces show that the SIP INVITE from the phone includes an offer with Opus and G.711ulaw, but the Room Kit's answer only includes G.711ulaw. Which statement explains why the Room Kit rejected the Opus codec?

Accepted Answer

The Room Kit's SIP profile is not configured to allow the Opus codec, so it rejected the offer.. The Room Kit's SIP profile determines which codecs it can use. If Opus is not enabled in the SIP profile, the Room Kit will not accept an offer that includes Opus, even if it supports the codec. The traces show the phone offered Opus, but the Room Kit answered with only G.711ulaw. Therefore, enabling Opus in the Room Kit's SIP profile is necessary to allow Opus negotiation.

Answer

The phone's SIP profile is configured to only offer Opus, and the Room Kit does not support it.

Answer

The region assigned to the phone does not include Opus, so CUCM did not offer it to the Room Kit.

Answer

The Room Kit does not support the Opus codec in its firmware version.

Question 9

A collaboration engineer is configuring a SIP trunk between Cisco Unified Communications Manager (CUCM) and a Cisco Unified Border Element (CUBE) for connectivity to an ITSP. The ITSP requires that all outbound calls use the G.729 codec to conserve bandwidth. However, the engineer notices that calls are being established with G.711. The engineer has already configured the SIP trunk to only allow G.729. Which additional configuration on CUCM is necessary to enforce G.729 for calls to the ITSP?

Accepted Answer

Set the 'Codec Preference' on the SIP trunk to G.729 only and ensure the Device Pool's Region uses G.729.. Codec enforcement on a SIP trunk to an ITSP requires aligning both the SIP trunk's codec preference and the Region configuration. The Region defines the codec policy for devices within and between regions. If the Region allows G.711, CUCM will include it in the offer, and the ITSP may select it. Therefore, setting the SIP trunk to G.729 only is insufficient; the Region must also be restricted to G.729 to ensure that only G.729 is offered.

Answer

Configure a Media Resource Group List (MRGL) with a hardware MTP that supports G.729.

Answer

Set the 'Codec Preference' in the Region configuration to G.729 only.

Answer

Set the 'Allowed Codecs' on the SIP Profile to G.729 only.

Question 10

A Cisco TelePresence SX80 codec is registered to CUCM as a SIP endpoint. Users report that when they dial the SX80 from a Cisco IP Phone 8861, the call connects but there is no audio. The SX80 is configured to use G.722 for all calls. The IP Phone 8861 supports G.722. Which CUCM configuration should you verify to ensure G.722 is used between these endpoints?

Accepted Answer

The region assigned to the SX80 and the IP Phone 8861 includes G.722.. Regions in CUCM control which codecs are available for calls between devices. If the region assigned to the SX80 and the IP Phone 8861 does not include G.722, the call will not use G.722, potentially causing no audio if the endpoints cannot agree on a codec. Verifying that the region includes G.722 and that both devices are assigned to it (or that inter-region settings permit G.722) is the correct action.

Answer

The device pool assigned to the SX80 has the 'G.722 Enabled' checkbox selected.

Answer

The SX80's SIP profile has the 'Enable G.722' option set to true.

Answer

The SIP profile assigned to the SX80 includes G.722 in the allowed codecs.

Question 11

A Cisco Webex Room 55 Dual is registered to CUCM using SIP. The device is not able to place calls to a third-party SIP server. You have verified that the SIP trunk to the third-party server is up and that the Room 55 is registered. Which TWO configuration elements on the Room 55 must be verified to ensure it can place calls through the SIP trunk? (Choose two.)

Accepted Answer

The Room 55's SIP profile has the correct 'SIP Transport' protocol (e.g., TCP, UDP) that matches the third-party server's requirements.. To place calls through a SIP trunk, the endpoint's codec support (via region) and SIP transport protocol must align with the third-party server's capabilities. The region ensures codec compatibility, and the SIP profile's transport setting ensures signaling compatibility. These two elements are critical for successful call setup.

Answer

The Room 55's SIP profile includes the correct 'Call Routing' settings to reach the third-party server.

Answer

The Room 55's SIP profile has the 'Outbound Proxy' setting configured to the third-party server's IP address.

Answer

The Room 55's SIP profile includes the correct SIP trunk destination address.

Question 12

A Cisco IP Phone 8861 is registered to CUCM and is using SIP. The phone is configured to use the G.729 codec for all calls. When the phone calls a Cisco Webex Room Kit that is also registered to CUCM, the call connects but the audio is garbled and choppy. The network administrator suspects a codec mismatch. Which CUCM configuration parameter should be checked first to ensure that both endpoints can negotiate a common codec?

Accepted Answer

The region assigned to both endpoints includes G.729 as an allowed codec.. The region configuration in CUCM determines which codecs are available for calls between devices. If G.729 is not included in the region assigned to both endpoints, CUCM will not offer it, leading to a codec mismatch and poor audio. Verifying that the region includes G.729 and that both endpoints are assigned to it (or that inter-region settings allow G.729) is the correct first step.

Answer

The device pool assigned to the IP Phone 8861 has the 'G.729 Enabled' checkbox selected.

Answer

The SIP profile assigned to the Room Kit includes the G.729 codec in its supported codec list.

Answer

The SIP profile assigned to the IP Phone 8861 includes G.729 in the allowed codecs list.

Question 13

A Cisco IP Phone 8845 registered to CUCM is configured to use only the G.722 codec for all calls. During a call with a Cisco Webex Room Kit that offers only G.711ulaw in its SDP, the phone displays 'Call Failed' and the call never connects. Which CUCM feature should the administrator configure to allow the call to succeed?

Accepted Answer

Add G.711ulaw to the phone's Advertised Codecs list in the device configuration.. The call fails because the phone advertises only G.722 while the Webex Room Kit offers only G.711ulaw, resulting in no common codec in the SDP negotiation. Adding G.711ulaw to the phone's Advertised Codecs list makes the phone offer a codec the Room Kit can accept, allowing the call to proceed. Transcoding and region settings cannot compensate for the absence of a mutually supported codec.

Answer

Enable 'Allow Transcoding' on the SIP trunk and register a transcoder with the G.722 and G.711ulaw codecs.

Answer

Configure a region pair with a maximum audio bit rate of 64 kbps between the phone's region and the Room Kit's region.

Answer

Enable 'Accept Unknown/Unsupported Codecs' on the SIP profile assigned to the phone.

Question 14

A collaboration engineer is troubleshooting a SIP call between a Cisco IP Phone 8845 and a Cisco Webex Room Kit registered to the same CUCM cluster. The engineer observes that the phone sends an INVITE with SDP containing a single media line for audio, but the Room Kit responds with a '488 Not Acceptable Here' message. The engineer verifies that both endpoints support G.711ulaw. Which statement explains the most likely cause?

Accepted Answer

The Room Kit requires that the INVITE include a media line for video, and the audio-only offer is rejected.. A '488 Not Acceptable Here' response often occurs when the SDP offer lacks a media type required by the terminating endpoint. The Webex Room Kit, as a video endpoint, may require a video media line in the offer. Since the phone sends an audio-only INVITE, the Room Kit rejects it. Ensuring the offer includes a video line, or configuring the Room Kit to accept audio-only calls, resolves the issue.

Answer

The phone's SIP profile does not have 'Early Offer' enabled, so the Room Kit rejects the delayed offer.

Answer

The Room Kit is configured to use only the Opus codec, and G.711ulaw is not supported in its SDP answer.

Answer

The SIP trunk between the phone and the Room Kit is configured with the wrong codec preference, causing the 488 response.

Question 15

A collaboration engineer is troubleshooting one-way audio on a call between a Cisco IP Phone 8845 registered to CUCM and a third-party SIP gateway. The SIP signaling completes successfully, but the caller cannot hear the called party. The engineer captures the SDP exchange and notices that the phone advertises its media address as a private 10.x.x.x address, while the gateway advertises a public IP address. Which action should the engineer take to resolve the one-way audio?

Accepted Answer

Enable Media Termination Point (MTP) on the SIP trunk to anchor the media and provide a public address for the SDP exchange.. One-way audio often occurs when an endpoint behind NAT advertises a private IP address in SDP, and the remote endpoint cannot send media to that address. Enabling Media Termination Point on the SIP trunk makes CUCM allocate a media resource that terminates the media and presents a reachable address in the SDP, ensuring bidirectional audio. This is the standard fix for NAT-related one-way audio in CUCM environments.

Answer

Configure a SIP normalization script on the SIP trunk to rewrite the SDP connection line with the public address of the CUCM server.

Answer

Configure the phone to use a public IP address by enabling NAT on the phone's network interface.

Answer

Modify the SIP trunk security profile to disable NAT and allow direct media between the endpoints.