The Expressway-C and Expressway-E pair can interconnect H 323-to-SIP calls for
voice.

Expressway-E provides a perimeter network that separates the enterprise network from
the Internet.

Summary:
Cisco Expressway is a solution for enabling secure, external collaboration. It consists of a pair of servers: Expressway-C (Core) inside the enterprise network and Expressway-E (Edge) in the DMZ. Its primary functions include providing secure Mobile and Remote Access (MRA) for endpoints, enabling Business-to-Business (B2B) calls, and performing protocol interworking between H.323 and SIP video devices.

Correct Option:

C. The Expressway-C and Expressway-E pair can interconnect H.323-to-SIP calls for voice and video:
This is a core function. The Expressway platform acts as a traversal zone and a gateway, providing protocol interworking. This allows legacy H.323 video endpoints (like TelePresence systems) to communicate with modern SIP-based endpoints, which is crucial for unified collaboration.

E. Expressway-E provides a perimeter network that separates the enterprise network from the Internet:
This is a foundational security principle of the design. Expressway-E is deployed in the DMZ (perimeter network), creating a secure boundary. It accepts external connections from the internet and acts as a controlled gateway to the internal Expressway-C server, which never faces the public internet directly.

Incorrect Option:

A. Expressway-E provides a VPN entry point for Cisco IP phones with a Cisco AnyConnect client...:
This is incorrect. Expressway MRA does not use a traditional VPN like AnyConnect. Instead, it uses a secure, firewall-friendly method based on TLS and HTTPS proxies to extend collaboration services. It does not require or use a VPN client.

B. Expressway-C provides encryption for Mobile and Remote Access but not for business-to-business communications:
This is incorrect. Encryption (TLS and SRTP) is a fundamental security feature that is applied to both MRA and B2B communication flows. The Expressway pair facilitates encrypted sessions for all traversal calls, regardless of whether they are from a remote employee or a business partner.

D. The Expressway-C and Expressway-E pair can enable connectivity from the corporate network to the PSTN via a T1/E1 trunk:
This is incorrect. The Expressway is not a PSTN gateway. Physical PSTN connectivity (like T1/E1 trunks or PRI) is handled by dedicated gateways (e.g., Cisco ISR G2 with PVDM), Cisco Unified Border Element (CUBE), or an SIP trunk to an ITSP. The Expressway's role is traversal and interworking, not direct PSTN termination.

Reference:
Cisco Expressway Series Deployment Guides (These guides detail the architecture, including the placement of Expressway-E in the DMZ and its functions for traversal, MRA, B2B, and protocol interworking).

G.711ulaw

Summary:
The SDP offer lists three possible audio codecs using their RTP/AVP (Audio/Video Profile) payload types: 0, 8, and 97. The Cisco IP Phone 7841, by default, supports a specific set of codecs. The phone will compare the offered codecs against its own capabilities and choose the first one it supports, in the order they are listed in the offer.

Correct Option:

A. G.711ulaw:
This is the correct answer. The payload types in the SDP offer correspond to:

0: G.711 µ-law

8: G.711 A-law

97: Dynamic range codec (not a standard audio codec like G.722)

The Cisco IP Phone 7841 supports G.711 µ-law by default. Since it is the first codec in the offer list (payload 0) that the phone supports, it will be selected for the call.

Incorrect Option:

B. iLBC:
The phone does not support the iLBC codec by default, and it is not represented by any of the payload types (0, 8, 97) listed in the SDP offer.

C. G.711alaw:
While the phone does support G.711 A-law (payload 8), it is listed after G.711 µ-law (payload 0) in the SDP offer. The phone will select the first mutually supported codec, which is G.711 µ-law.

D. G.722:
The payload type for G.722 is 9, which is not present in the offer. Payload type 97 is not G.722; it is typically used for a dynamic payload type that would be defined later in the SDP (e.g., with an a=rtpmap attribute), which is not shown here. The 7841 also does not support the wideband G.722 codec.

Reference:
Cisco Unified Communications Manager System Guide, Codec Configuration (This guide details how codec negotiation works and lists the default capabilities of various phone models, confirming the 7841's support for G.711).

Inbounds CSS on any gateway typically should have access to internal destinations only and not PSTN destinations.

Classify all route patterns as on-net or off-net and prohibit off-net call transfers in Cisco
Unified CM Service parameters.

Summary:
Toll fraud occurs when unauthorized users gain access to a company's phone system to make unauthorized PSTN calls. To mitigate this, the core principle is to restrict the PSTN destinations that can be reached from vulnerable entry points, such as gateways. This involves carefully controlling Call Search Spaces (CSS) and limiting the ability to transfer calls to external numbers.

Correct Option:

A. Inbounds CSS on any gateway typically should have access to internal destinations only and not PSTN destinations:
This is a primary defense. The Calling Search Space (CSS) assigned to the inbound leg of a gateway (often via a Route Pattern or Trunk configuration) should be highly restrictive. It should only contain Partitions with internal Directory Numbers (DNs), preventing an outside caller from dialing back out to the PSTN through the same system, a common tactic in fraud.

C. Classify all route patterns as on-net or off-net and prohibit off-net call transfers in Cisco Unified CM Service parameters:
This creates a critical transfer barrier. By marking PSTN route patterns as "Off-Net" in the Class of Service control settings, you can then use the service parameter "Block OffNet to OffNet Transfer" to prevent a common fraud scenario where an attacker on a PSTN call transfers it to another expensive PSTN number, making the company pay for both legs of the call.

Incorrect Option:

B. Classify all route patterns as on-net and prohibit on-net to on-net call transfers:
This is incorrect and counterproductive. Classifying all patterns as "on-net" would include PSTN routes, which is the opposite of what is needed. Furthermore, blocking on-net to on-net transfers would severely disrupt internal business operations by preventing basic call transfers between colleagues.

D. Inbound CSS on any gateway typically should have access to internal destinations and PSTN destinations:
This is the direct cause of many toll fraud incidents. Giving an inbound gateway's CSS access to PSTN destinations allows an attacker to use your system as a free dial-through portal, running up massive toll charges.

E. Classify all route pattern as off-net and prohibit off-net to off-net call transfers:
While prohibiting off-net to off-net transfers is good, classifying all route patterns as off-net is incorrect. Internal route patterns for inter-office calls must be classified as "on-net" to function correctly and to allow the service parameter to distinguish between internal and external calls.

Reference:
Cisco Unified Communications Manager Security Guide, Toll Fraud Prevention (This official guide details these exact best practices, including restricting gateway inbound CSS and using the "Block OffNet to OffNet Transfer" service parameter).

Add a transcoder that supports G.711ualw and G.729.

Summary:
The SIP INVITE contains an SDP offer with a media line (m=audio) that lists a single codec: G.711 µ-law, indicated by payload type 0 (RTP/AVP 0). The problem states that the endpoint receiving this INVITE only supports G.729. Since there is no overlap between the offered codec (G.711ulaw) and the endpoint's capabilities (G.729), the call will fail as there is no common codec for the media path.

Correct Option:

B. Add a transcoder that supports G.711ulaw and G.729:
This is the correct solution. A transcoder is a DSP resource that converts one codec to another. By placing a transcoder in the media path, it can accept the G.711ulaw stream from the caller, transcode it to G.729, and then send it to the callee (and vice-versa). This provides the necessary codec compatibility for the call to succeed.

Incorrect Option:

A. Nothing; both sides support G.729:
This is incorrect. The SDP offer explicitly shows only payload type 0 (G.711ulaw). There is no mention of payload type 18 (G.729) in the offer, meaning the caller does not support it. Therefore, both sides do not support G.729.

C. Add a media termination point that supports G.711ulaw and G.729:
A Media Termination Point (MTP) is used to provide media services like conferencing, annunciators, or to handle RFC 2833 DTMF packets. While some MTPs can transcode, their primary purpose is not codec conversion. A transcoder is the specific and correct resource designed for this task.

D. Nothing; both sides support payload type 101:
This is incorrect. Payload type 101 is a legacy dynamic payload type often used for telephone-event (RFC 2833 DTMF), not for primary audio. The SDP offer does not list payload type 101, and even if it did, it is not a voice codec and cannot be used to carry the audio conversation.

Reference:
co Unified Communications Manager System Guide, Media Resources (This guide explains the different media resource types, clarifying the role of a transcoder versus an MTP).

In CUCM Service Parameters set “Opus codec Enables” to “Enabled for all devices
Except Recording-Enabled devices.

Summary:
The OPUS codec is a high-quality, low-bandwidth audio codec. However, many call recording solutions cannot decode or transcribe it, leading to silent or unrecorded calls. To prevent this, Cisco UCM provides a centralized service parameter that controls OPUS advertisement specifically for devices that are marked as "Recording Enabled," ensuring compatibility with recording platforms.

Correct Option:

A. In CUCM Service Parameters set “Opus codec Enables” to “Enabled for all devices Except Recording-Enabled devices.”
This is the correct and direct method. The service parameter "Opus codec Enable" in the "Clusterwide Parameters (Device - SIP)" section offers this exact option. When set this way, UCM will advertise OPUS to standard phones but will exclude it from the SDP offered to any device that has its "Recording" flag enabled, thus ensuring recording compatibility.

Incorrect Option:

B. Go to the phone’s configuration page and set “Advertise OPUS Codec” to be “false”.
This is incorrect because there is no such per-phone configuration setting for the OPUS codec on the phone configuration page in Cisco UCM. The control is implemented globally via a service parameter.

C. Route recorded calls through Cisco Unified Border Element because it does not support OPUS.
This is an inefficient workaround, not a solution. While a CUBE could be configured to transcode the codec, it adds unnecessary complexity and a point of failure. The correct method is to prevent UCM from offering OPUS in the first place using its built-in mechanism.

D. Integrate the Cisco Unified CM with a recording solution that does not support OPUS.
This is not a proactive configuration change on UCM. Forcing a change to the entire recording infrastructure is a costly and disruptive alternative to simply changing a single service parameter in UCM.

Reference:
Cisco Unified Communications Manager Administration Guide, Service Parameters Configuration (This guide details the cluster-wide service parameters, including the "Opus codec Enable" parameter and its options for managing advertisement to recording-enabled devices).

Cisco Unified IP phones (all models)

SIP trunks

Summary:
The Bulk Administration Tool (BAT) in Cisco Unified Communications Manager (UCM) is designed for mass operations on specific types of objects. Its primary function is to add, update, or delete a large number of similar devices or configuration elements simultaneously. It is optimized for common, standardized endpoints and trunks, not for every single object type in the system.

Correct Option:

A. Cisco Unified IP phones (all models):
This is a primary function of BAT. It is extensively used to add, update, or delete large numbers of IP phones (both SCCP and SIP) by importing data from a CSV file, which is essential for initial deployments or large-scale changes.

E. SIP trunks:
BAT supports the bulk administration of SIP trunks. This allows administrators to create, update, or delete multiple SIP trunk configurations in a single operation, streamlining the setup of connectivity to ITSPs or other SIP systems.

Incorrect Option:

B. H.323 clients:
BAT does not support H.323 clients. H.323 clients (like the old Cisco IP VCphone) are managed individually through the UCM administration interface and are not available for bulk operations in the BAT.

C. H.225 trunks:
While BAT supports SIP trunks, it does not support H.225 trunks. H.323/H.225 trunk configuration is handled on a per-trunk basis within the UCM admin interface and is not available as a template-driven bulk operation.

D. Music on hold servers:
Music on Hold (MOH) servers are a service-level configuration in UCM. There is typically only one or a few MOH servers per cluster, so there is no need for a bulk administration tool to manage them. They are configured individually.

Reference:
Cisco Unified Communications Manager Bulk Administration Guide (This official guide lists the specific device types and features that are supported by the BAT, confirming phones and SIP trunks as key supported items).

Add the requirements for each audio and video codec and how many calls must be
supported.

Summary:
Location-Based Call Admission Control (CAC) in Cisco Unified Communications Manager (UCM) manages the bandwidth for calls between locations to prevent oversubscription of WAN links. The bandwidth requirement for a location link is not automatically discovered; it must be manually calculated and configured by an engineer based on the expected call volume and the bandwidth consumed by each type of call.

Correct Option:

D. Add the requirements for each audio and video codec and how many calls must be supported:
This is the correct method. The engineer must:

Identify the audio and video codecs used for intra-cluster calls (e.g., G.711, G.729, G.722, H.264).

Determine the total bandwidth per call for each codec (including IP overhead).

Decide the maximum number of concurrent calls of each type that the link should support.

Sum the total bandwidth:
(Bandwidth per G.711 call * Max G.711 calls) + (Bandwidth per G.729 call * Max G.729 calls) + (Bandwidth per Video call * Max Video calls) = Total CAC Bandwidth Requirement.

Incorrect Option:

A. Execute the Resource Reservation protocol to return location-based requirements:
This is incorrect. RSVP is a protocol used for dynamic CAC on a per-call basis (e.g., between a UCM cluster and a Cisco Unified Border Element). It is not a tool for determining the static, total bandwidth capacity that should be configured for a UCM Location.

B. Calculate the number of calls against the license for Cisco Unified Border Element to determine calls per location:
This is incorrect. CUBE licensing is based on the number of concurrent sessions and is unrelated to the internal bandwidth calculations between UCM Locations. The two concepts are separate.

C. Set the requirements in the service parameters:
This is incorrect. While there are service parameters for CAC, they are for enabling/disabling the feature or setting system-wide limits (like the default audio/video bandwidth for regions). The specific bandwidth for each Location is configured directly within the Location configuration itself, not in a global service parameter.

Reference:
Cisco Unified Communications Manager System Guide, Call Admission Control (This guide explains the purpose of locations and regions, detailing how bandwidth is managed and must be configured by the administrator).

SIP/SIMPLE

Summary:
Cisco Jabber is a unified communications client that provides instant messaging (IM) and presence. The underlying protocol used for these specific features depends on the deployment mode. When Jabber is registered to Cisco Unified Communications Manager (UCM), it uses the SIP/SIMPLE protocol for IM and presence, leveraging the built-in capabilities of the UCM IM and Presence Service.

Correct Option:

C. SIP/SIMPLE:
This is the correct answer. SIP/SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) is a set of standards that extends the core SIP protocol to support instant messaging and presence information. The Cisco IM and Presence Service, which is integrated with UCM, uses SIP/SIMPLE as its native protocol for communication between Jabber clients and the service.

Incorrect Option:

A. Jabber:
"Jabber" is the name of the client application and the overall brand, not the name of the underlying communication protocol it uses.

B. P2P (Peer-to-Peer):
This is incorrect. Cisco Jabber does not use a direct P2P protocol for its core IM and presence services. All communication is typically centralized through the IM and Presence Service nodes for security, logging, and compliance reasons.

D. XMPP (Extensible Messaging and Presence Protocol):
While XMPP is a standard protocol for IM and presence, it is not the primary protocol used by Jabber in a UCM-integrated deployment. Cisco primarily uses SIP/SIMPLE for its native on-premises IM and Presence Service. XMPP is more commonly associated with public services like Google Talk or used for federation with other platforms.

Reference:
Cisco Jabber Planning Guide, Protocol Support (This planning guide details the architecture and confirms that the on-premises deployment of Jabber uses the Cisco IM and Presence Service, which is built on SIP/SIMPLE).

The transcoder resource is missing.

Summary:
The conference bridge at the main site is configured to use only the G.711 codec. The remote user's phone only supports the G.729 codec. For the conference to be established, the audio streams from all participants must be compatible with the conference bridge. Since there is a codec mismatch between the remote user (G.729) and the conference bridge (G.711), a media resource is required to convert between these codecs.

Correct Option:

C. The transcoder resource is missing:
This is the direct cause of the issue. A transcoder is a DSP-based media resource that performs codec conversion. In this scenario, a transcoder is needed to convert the remote user's G.729 audio stream into G.711 for the conference bridge to process, and vice-versa. The error indicates that UCM could not find an available transcoder resource to perform this necessary conversion, preventing the conference from being established.

Incorrect Option:

A. The remote phone does not have the conference feature assigned:
While this could cause a conference failure, the error message and scenario point to a media negotiation issue, not a feature license one. The problem is specifically the codec mismatch after the user attempted to initiate the conference.

B. A Media Termination Point is missing:
An MTP is used for other media services, such as providing hold tones or handling DTMF, but its primary role is not codec transcoding. While some advanced MTPs can transcode, the dedicated and correct resource for this task is a transcoder.

D. A software conference bridge is not assigned:
The problem states a conference bridge is configured. The issue is not the absence of a bridge, but the incompatibility between the remote user's codec and the codec the bridge requires. Assigning a different software conference bridge that also only supports G.711 would not resolve the core codec mismatch.

Reference:
Cisco Unified Communications Manager System Guide, Conference Bridges (This guide explains how conference bridges operate and the need for compatible codecs or transcoding resources when participants use different codecs).

Cisco UCM class of service

Summary:
Toll fraud prevention in Cisco Unified Communications Manager (UCM) is fundamentally about controlling which users and devices are authorized to make specific types of calls, especially expensive PSTN calls. The primary tool for implementing this authorization is a logical grouping of route patterns with associated permissions.

Correct Option:

A. Cisco UCM class of service:
This is the correct and foundational tool. Class of Service (CoS) in UCM is implemented through the combination of Partitions and Calling Search Spaces (CSS). This is the primary mechanism to begin implementing toll fraud prevention. You create Partitions to group route patterns (e.g., "Local-PSTN," "International-PSTN") and assign CSS to devices/users to control which partitions they can access. This directly prevents unauthorized PSTN dialing.

Incorrect Option:

B. Cisco Unified Mobility:
This feature allows a user to have a single number that can ring multiple devices (like a mobile phone). While it has security considerations, it is not the primary tool used to build a general toll fraud prevention policy for the entire system.

C. Cisco UCM Access Control Group restrictions:
This is a distractor. While "Access Control Groups" exist for specific features like Extension Mobility, the universal access control model for call routing in UCM is based on Partitions and Calling Search Spaces, not a feature with this exact name.

D. Cisco Unified Real-Time Monitoring Tool:
This is a monitoring and reporting tool used for troubleshooting and viewing real-time system data. It is an analysis tool, not a configuration tool used to implement security policies like toll fraud prevention.

Reference:
Cisco Unified Communications Manager System Guide, Call Routing and Authorization (This guide details the core concepts of Partitions and Calling Search Spaces, which form the basis of the Class of Service model for controlling call authorization).

show policy-map interface GigabitEthernet 1/0/1

Summary:
After implementing a Quality of Service (QoS) policy on a Cisco switch interface, an engineer needs to verify that the policy is active and functioning correctly. This involves confirming that traffic is being classified into the correct classes and that the appropriate markings (DSCP, CoS) are being applied as packets are processed by the interface.

Correct Option:

D. show policy-map interface GigabitEthernet 1/0/1:
This is the definitive command for verification. It displays real-time statistics for a service policy applied to a specific interface. The output shows:

Which class-map each packet matched.

The number of packets/bytes that matched each class.

The policing or marking actions that were taken.

This provides direct evidence of correct classification and marking.

Incorrect Option:

A. show policy-map:
This command only displays the configuration of the policy maps themselves (the "what should happen"). It does not show whether the policy is applied to an interface or provide any live statistics about its operation.

B. show class-map interface GigabitEthernet 1/0/1:
This is not a valid Cisco IOS command. The show class-map command exists, but it only shows the class-map configuration, not interface-specific statistics.

C. show access-lists:
This command displays the configuration and hit counts for standard and extended IP access control lists (ACLs). While ACLs can be used within class-maps for classification, this command alone does not show the overall QoS policy operation, marking results, or which policy is applied to an interface.

Reference:
Cisco IOS QoS Configuration Guide, Monitoring QoS (This official guide details the commands used to monitor and troubleshoot QoS, confirming that show policy-map interface is the primary tool for verifying service policy statistics).

Check for duplex/speed mismatches between the network port settings of the system
and network switch.

Summary:
The symptoms of bad video quality and choppy voice are classic signs of network issues, specifically packet loss, jitter, or latency. Since changing the device pool (which controls regions, locations, and other UCM settings) did not resolve the issue, the problem is almost certainly not a Call Admission Control (CAC) or UCM configuration problem. The focus must shift to the underlying physical and data link network connectivity.

Correct Option:

B. Check for duplex/speed mismatches between the network port settings of the system and network switch:
This is a fundamental and common troubleshooting step. A duplex mismatch (e.g., one side set to full duplex, the other to auto/half) causes collisions and frame errors on the link, leading to severe packet loss. This directly results in the choppy audio and poor video quality described. Verifying and ensuring consistent speed and duplex settings is a critical first action.

Incorrect Option:

A. Set the service parameter Use Video Bandwidth Pool for Immersive Video Calls to "false":
This parameter is related to bandwidth accounting for specific immersive TelePresence systems, not the general video endpoints experiencing this issue. It is unrelated to the fundamental media quality problems being reported.

C. Restart the Cisco Location Bandwidth Manager service on the Cisco UCM publisher:
This service manages CAC. Since changing the device pool (and thus the associated Location and Region) did not help, the issue is not with CAC being incorrectly calculated or enforced. Restarting the service would not resolve a underlying network problem causing packet loss.

D. Use direct IP address calls between two endpoints to troubleshoot call quality issues:
While this can be a valid test to bypass UCM call routing, it does not isolate the root cause. If the problem is a local duplex mismatch or a bad switch port, the direct IP call will still traverse the faulty network segment and exhibit the same poor quality. It is a diagnostic step, but checking the physical/link layer configuration is a more direct and fundamental first action.

Reference:
Cisco Collaboration System Solution Reference Network Designs (SRND), Network Infrastructure (This design guide emphasizes the critical importance of proper network configuration, including specifying fixed speed and duplex settings for all collaboration endpoints to prevent performance issues).