500-710 Practice Exam Questions with Answers Video Infrastructure Implementation (VII) Certification

 Microsoft Skype for Business (S4B) interoperability with Cisco Meeting Server (CMS) allows S4B users to join CMS meetings and vice versa. Chat can be used between S4B and Cisco Meeting App (CMA) users, as both support the SIP SIMPLE protocol for instant messaging. However, chat cannot be used between S4B and Cisco Meeting Server Spaces, as Spaces do not support chat functionality. Calls can be placed between S4B and CMA users, as both support the H.264 video codec and the Opus audio codec. Local directories can be used between S4B users and CMA, but only if the CMA is configured to use the same domain as the S4B server. References: Configure Cisco Meeting Server and Skype for Business, Plan for Video Interop Server in Skype for Business Server, Deploy Video Interop Server in Skype for Business Server

• n endpoint is a device that can make or receive calls, such as a video conferencing system, a phone, or a soft client1.
• An endpoint can register to an Expressway to obtain a SIP address and a directory number, and to enable call routing and provisioning2.
• To view a list of all endpoints registered to an Expressway, an engineer can go to the web interface of the Expressway and navigate to Status > Registrations > By Device3. This page shows the following information for each registered endpoint4:
• The other options are incorrect because:

References := 1: Cisco Expressway Basic Configuration Deployment Guide (X8.11) 2: Cisco Expressway Administrator Guide (X12.6) 3: Cisco Expressway Web Interface Reference Guide (X12.6) 4: Cisco Expressway and CUCM via SIP Trunk Deployment Guide (CUCM 8,9 and X8.1.1) : Cisco Expressway Command Line Interface Reference Guide (X12.6) : Cisco Expressway Series - Troubleshooting Call History

According to the Cisco documentation1, the engineer can look on the Conductor web interface under Status > Calls > In Progress to view information about a call currently in progress. This page shows the details of each call, such as the call ID, the conference alias, the call type, the call state, the call duration, the call rate, the call protocol, the encryption status, and the call participants. The other options, such as Status > Calls > Calls, Logs > Event Log, and Status > Conferences > In Progress, do not show the information about a call currently in progress, but rather show the information about all calls, the event log messages, and the conferences in progress, respectively. References: 1: Cisco TelePresence Conductor Administrator Guide (XC4.3), Chapter 5: Monitoring Conferences and Calls, https://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/conductor/admin_guide/Cisco-TelePresence-Conductor-Administrator-Guide-XC4-3.pdf

= SIP (Session Initiation Protocol) is the protocol that is used for VoIP and other multimedia applications. SIP uses port 5060 for non-encrypted signaling traffic and port 5061 for traffic encrypted with Transport Layer Security (TLS). These are the ports that SIP clients typically use to communicate with SIP servers and other endpoints1234 References := 1Session Initiation Protocol - Wikipedia 2SIP Port Numbers used by Providers - WhichVoIP 3What is a SIP port? | FREEZVON 4An Introduction to SIP Protocol: Definition, Features, & More - Nextiva

 The Cisco Meeting Server uses port 5060 for SIP signaling from the call bridge to the H.323 gateway. This is because port 5060 is the default port for SIP signaling over UDP or TCP, and the Cisco Meeting Server supports both protocols for SIP signaling1. The call bridge communicates with the H.323 gateway using SIP signaling to establish and manage calls between SIP and H.323 endpoints2.

The other options are not correct because:

• 4999: This is not a port used for SIP signaling, but for H.323 signaling over TCP. The Cisco Meeting Server uses port 4999 for H.323 signaling from the call bridge to the H.323 gateway1.
• 5222: This is not a port used for SIP signaling, but for XMPP signaling over TCP. The Cisco Meeting Server uses port 5222 for XMPP signaling from the call bridge to the XMPP server1.
• 6061: This is not a port used for SIP signaling, but for SIP signaling over TLS. The Cisco Meeting Server uses port 6061 for SIP signaling over TLS from the call bridge to the Expressway-C or the SIP proxy1.

References := 1: Cisco Meeting Server 3.5 Configuration Guides - Port Reference 2: Cisco Meeting Server 3.5 Configuration Guides - H.323 Gateway Configuration

 = The Cisco event log on the Telepresence Server shows key events that have occurred on the server, such as call events, login events, and alarms. The event log can be accessed through the web interface or the command-line interface (CLI) of the server. The event log allows the administrator to view and sort through conferencing events, such as conference creation, deletion, joining, leaving, and media negotiation. The event log also indicates the status and outcome of each event, such as success, failure, or error. The event log can be useful for troubleshooting and monitoring purposes123 References := 1Cisco TelePresence Server Administration and User Guide (4.4) - Event log 2Cisco TelePresence Server Command Reference Guide (4.4) - file view log 3Cisco TelePresence System Message Overview

= Call-control devices are responsible for establishing, maintaining, and terminating calls in a Cisco Collaboration environment. They provide features such as endpoint registration, call routing, call admission control, codec negotiation, and trunk protocol translation. According to the Cisco Expressway Basic Configuration Deployment Guide, Cisco Expressway is a call-control device that enables secure communication between endpoints inside and outside the enterprise network. According to the Call Control with CUCM for Cisco CICD (210-060) course, Cisco Unified Communications Manager is a call-control device that provides centralized call processing and management for Cisco IP phones, video endpoints, and gateways.

The other options are not call-control devices, but rather collaboration devices that provide different functions. Cisco Telepresence Endpoint is a video device that enables face-to-face communication with high-quality audio and video. Cisco Meeting Server is a conferencing device that enables multiparty audio, video, and web collaboration. Cisco Content Server is a recording and streaming device that enables live and on-demand video content creation and distribution.

References := Cisco Expressway Basic Configuration Deployment Guide, Call Control with CUCM for Cisco CICD (210-060), [Cisco Collaboration Devices At-a-Glance]

• Authentication credentials can be stored in the local database of the Expressway. This is true because the Expressway can use its local authentication database to verify the credentials of incoming requests from devices or external systems1. Alternatively, the Expressway can also use external authentication methods, such as LDAP, SAML, or OAuth1.
• SIP endpoints always need authentication credentials. This is false because the Expressway’s authentication policy can be configured separately for each zone, and it can be set to “Do not check credentials” or “Treat as authenticated” for SIP messages from local or non-local domains1. This means that some SIP endpoints may not need authentication credentials to communicate with the Expressway, depending on the zone configuration.
• The endpoint and Expressway must be configured and synchronized with an NTP server delivering an identical time stamp. This is false because the Expressway and the endpoint do not need to be synchronized with identical NTP timestamps for authentication purposes2. However, it is recommended to synchronize the Expressway and the endpoint with the same NTP server for other reasons, such as logging, troubleshooting, and certificate validation3.
• Expressways and endpoints do not need to be synchronized with identical NTP timestamps. This is true because the Expressway and the endpoint do not need to be synchronized with identical NTP timestamps for authentication purposes2. However, it is recommended to synchronize the Expressway and the endpoint with the same NTP server for other reasons, such as logging, troubleshooting, and certificate validation3.
• When a subzone is set for “Treat as Authenticated”, the endpoint must have the correct authentications configured. This is false because the “Treat as Authenticated” option means that the Expressway will not verify the credentials and will allow the message to be processed as if it has been authenticated1. This option can be used to cater for endpoints from third-party suppliers that do not support authentication within their registration mechanism1.

References := 1: Cisco Expressway Administrator Guide (X14.0) - Device Authentication 2: Cisco Expressway Administrator Guide (X14.0) - NTP Configuration 3: Cisco Expressway Administrator Guide (X14.0) - Time and Date

When a source endpoint dials a destination endpoint, the first thing to occur is that an invite is sent to the SIP Server. The SIP Server is a logical entity that handles the signaling and call setup between SIP endpoints. The invite message contains the source endpoint’s identity, capabilities, and session parameters, as well as the destination endpoint’s address. The SIP Server thenroutes the invite message to the destination endpoint or to another SIP Server if the destination endpoint is in a different domain12. References := 12

 Cisco WebEx Meeting Center is a cloud-based service that provides video conferencing and collaboration features for WebEx clients and endpoints. It supports calls from Cisco IP phones, Cisco Jabber, Cisco TelePresence, and third-party SIP and H.323 devices. It also integrates with Cisco Unified Communications Manager (UCM) and Cisco Expressway for call control and routing. Cisco WebEx Meeting Center allows users to schedule, join, and host meetings from any device, and share content, chat, and record sessions. It also offers artificial intelligence-powered features such as noise removal, transcription, and translation. References: Cisco WebEx Meeting Center Data Sheet, Deployment guide for Calling in Webex App (Unified CM), Webex Calling for Microsoft Teams

= Subzones are logical groupings of endpoints that are registered to an Expressway. Subzones are used for the following purposes1:

• Endpoints are assigned to a particular subzone by their alias or IP subnet: This is true, as the Expressway can use either the alias pattern or the IP subnet of the endpoint to determine which subzone it belongs to. The alias pattern is a regular expression that matches the endpoint’s SIP URI or H.323 ID. The IP subnet is a range of IP addresses that the endpoint falls into2.
• Subzones are used for bandwidth and authentication control of a logical grouping of endpoints within the registration list of the Expressway: This is true, as the Expressway can apply different bandwidth limits and authentication policies to different subzones. The bandwidth limits can be set for inbound and outbound calls, as well as for inter-subzone and intra-subzone calls. The authentication policies can be set to require or not require credentials from the endpoints in a subzone3.

The other options are not correct because:

• Subzones require the use of endpoint authentication: This is not true, as the Expressway can configure subzones to not require authentication from the endpoints. This can be useful for allowing guest or anonymous access to the video network3.
• Subzones are used for routing calls to peers in other parts of the network: This is not true, as the Expressway uses zones, not subzones, for routing calls to peers in other parts of the network. Zones are logical connections between Expressways or other SIP or H.323 entities, such as gateways, proxies, or servers. Zones can have different types, such as traversal, neighbor, or DNS4.
• Endpoints require the use of other endpoints for authentication: This is not true, as the Expressway does not use endpoints for authentication, but rather uses its own internal database or an external LDAP server. The Expressway can also use certificates or tokens for authentication, but these are not provided by other endpoints5.

References := 1: Cisco Expressway Basic Configuration Deployment Guide (X12.6) - Subzones [Cisco Expressway Series] - Cisco 2: Cisco Expressway Basic Configuration Deployment Guide (X12.6) - Assigning Endpoints to Subzones [Cisco Expressway Series] - Cisco 3: Cisco Expressway Basic Configuration Deployment Guide (X12.6) - Configuring Subzone Bandwidth and Authentication [Cisco Expressway Series] - Cisco 4: Cisco Expressway Basic Configuration Deployment Guide (X12.6) - Zones [Cisco Expressway Series] - Cisco 5: Cisco Expressway Basic Configuration Deployment Guide (X12.6) - Authentication [Cisco Expressway Series] - Cisco

 A SIP alias is an alternative address for a SIP URI, which is the standard format for identifying a user or a service in a SIP network. A SIP URI consists of a user part, an at sign (@), and a host part, such as user@host. The user part can be a username, a phone number, or any other identifier. The host part can be a domain name, an IP address, or any other network address. For example, alice@example.com, +1234567890@voip.com, and bob@192.168.1.1 are valid SIP URIs1.

A SIP alias is a way of creating multiple SIP URIs for the same user or service, using different user parts or host parts. For example, if a user has a SIP URI of john.doe@cisco.com, he can also have a SIP alias of jdoe@cisco.com, or john.doe@voip.cisco.com, or any other combination of user and host parts. A SIP alias can be useful for providing different contact options, simplifying dialing, or hiding the real SIP URI for privacy or security reasons2.

To configure a SIP alias, you need to have a SIP account with a SIP service provider that supports SIP alias, such as iptel.org. You also need to have a SIP device or application that can register and use the SIP alias, such as a VoIP phone, a softphone, or a router. You can create and manage your SIP alias using the web interface of your SIP service provider, or using the configuration settings of your SIP device or application2.

The correct format for a SIP alias is the same as the format for a SIP URI, which is user@host. Therefore, the option C. John.doe@cisco.com is the correct format for a SIPalias, as it has a user part (John.doe) and a host part (cisco.com) separated by an at sign (@). The other options are not correct formats for a SIP alias, because:

• A. 212-555-1212 : This is a phone number, not a SIP URI. It does not have a user part or a host part, and it does not have an at sign (@).
• B. +12125551212 : This is also a phone number, not a SIP URI. It does not have a user part or a host part, and it does not have an at sign (@).
• D. John.doe : This is only a user part, not a SIP URI. It does not have a host part, and it does not have an at sign (@).

References := Session Initiation Protocol (SIP) Parameters, How to configure SIP Alias | DrayTek

The SIP message that is sent from the called endpoint after the call request is accepted is “200 OK”. This message indicates that the called endpoint has agreed to participate in the session and is ready to receive media. The “200 OK” message also contains the session description protocol (SDP) information that specifies the media capabilities and parameters of the called endpoint, such as the codec, the port number, and the encryption method. The “200 OK” message is sent as a response to the “INVITE” message that was sent by the caller to initiate the session. The caller then acknowledges the “200 OK” message by sending an “ACK” message, which completes the session establishment12

The other options are not valid SIP messages that are sent from the called endpoint after the call request is accepted. “Hello” is not a SIP message at all, but a common greeting. “404 OK” is a nonsensical combination of a SIP error code and a SIP status code. “404” means “Not Found”, which indicates that the requested user or service does not exist or is not available. “OK” means “Success”, which indicates that the request was fulfilled. These two codes are contradictory and cannot be used together. “Connect” is not a SIP message either, but a generic term for establishing a link or a communication34 References: SIP - Basic Call Flow - Online Tutorials Library, Understanding SIP INVITE method and messages - Wildix blog, Session Initiation Protocol - Wikipedia, Basic SIP Call Flows & Troubleshooting Commands - Cisco Community

• Hybrid Services are a set of features that allow you to connect your on-premises or partner-hosted Cisco collaboration components to the cloud-based Cisco Spark capabilities1.
• Connectors are software modules that run on Expressway-C or Expressway-E and provide secure communication between your premises environment and the cloud1.
• To enable Hybrid Services with Cisco Spark, you need to establish four connectors in your premises environment123:
• The other options are incorrect because:

References := 1: The best of both worlds – Cisco Spark Hybrid Services 2: Deployment guide for Hybrid Calendar - Deploy Expressway … - Cisco 3: Deployment guide for Hybrid Message - Deploy Cisco Webex Hybrid Message … : Deployment guide for Serviceability Connector - Deploy … - Cisco : Deployment guide for Hybrid Calling for Webex Devices (Device … - Cisco

• The Cisco TelePresence Conductor is a conference resource manager that orchestrates and allocates conference bridges and media resources for scheduled and ad hoc conferences1.
• To collect information on system activity, including TCP dumps, of the Conductor over a period of time, an engineer can use the diagnostic logging feature of the Conductor2. Diagnostic logging allows the engineer to capture detailed logs and network traces for troubleshooting and analysis2.
• To enable diagnostic logging, the engineer can go to the web interface of the Conductor and navigate to Maintenance > Diagnostics > Diagnostic Logging2. The engineer can then select the components and levels of logging, the duration and frequency of logging, and the destination of the log files2.
• The other options are incorrect because:

References := 1: Cisco TelePresence Conductor Product Overview 2: Cisco TelePresence Conductor Administrator Guide (XC4.3) 3: Cisco TelePresence Conductor Web Interface Reference Guide (XC4.3) : Cisco TelePresence Conductor Network Log Messages (XC4.3) : Cisco TelePresence Conductor CDR Data Dictionary (XC4.3)

= The Calendar Connector for Cisco Spark Hybrid is a feature that allows users to schedule and join Webex meetings from their Microsoft Exchange calendars. To integrate the Calendar Connector with Microsoft Exchange, an impersonation account is needed on the Exchange server. An impersonation account is a service account that has the permission to access other users’ calendars and send meeting invitations on their behalf. The Calendar Connector uses the impersonation account to monitor and update the users’ calendars, and to add Webex meeting details to the invitations. To create and configure an impersonation account on Microsoft Exchange, follow the steps in the deployment guide1. References := Deployment guide for Hybrid Calendar - Deploy Expressway-based Hybrid Calendar for Microsoft Exchange

Interoperability is a method for controlling bandwidth within a subzone on the Expressway. Interoperability allows the Expressway to adjust the bandwidth of calls based on the capabilities and preferences of the endpoints involved. For example, if a SIP endpoint calls an H.323 endpoint, the Expressway can downspeed the call to match the lower bandwidth of the H.323 endpoint. Interoperability also enables the Expressway to interwork different protocols, such as SIP and H.323, and different media formats, such as H.264 and H.263. Interoperability can be configured on the Expressway by setting the interoperability mode and the interoperability profile for each subzone. The interoperability mode determines whether the Expressway will try to interwork calls within the subzone or not. The interoperability profile defines the parameters for interworking, such as the preferred protocol, the preferred media format, and the maximum bandwidth. To configure interoperability on the Expressway, go to Configuration > Local Zone > Default Subzone or Configuration > Local Zone > Subzones and select the subzone you want to modify. Then, go to the Interoperability section and select the interoperability mode and profile for the subzone. References := Cisco Expressway Administrator Guide (X15.0) - InteroperabilityCisco Expressway Basic Configuration Deployment Guide (X15.0) - Configuring Interoperability

According to the Cisco documentation1, the Expressway stores the username and password of the local database users in its local database. These users can be used for authentication and authorization purposes when accessing the Expressway web interface or API. The other elements, such as user authority, CA authority, and crypto hash key, are not stored in the local database of the Expressway, but are either derived from other sources or generated dynamically. References: 1: Cisco Expressway Basic Configuration Deployment Guide, Chapter 3: Configuring Local Database Users, https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-11/Cisco-Expressway-Basic-Configuration-Deployment-Guide-X8-11.pdf

= To schedule a CMS conference on TMS with WebEx as a participant, the engineer needs to ensure that the WebEx server address is correctly configured in both TMS and CMS. This is because TMS uses the WebEx server address to create an externally hosted conference with a WebEx SIP URI, and CMS uses the WebEx server address to dial out to the WebEx meeting. The WebEx server address can be configured in TMS under Administrative Tools > Configuration > Network Settings > WebEx Settings, and in CMS under Configuration > WebEx. The WebEx server address should match the WebEx site URL that is used to join the WebEx meeting12. References := 12

• The SIP INVITE message is a request that initiates a session between two or more SIP endpoints1.
• The SIP INVITE message contains a Request-URI, which specifies the destination address of the session1.
• The Request-URI can be either a SIP URI or a tel URI1. A SIP URI has the format sip:user@host, where user is the username or phone number, and host is the domain name or IP address of the SIP server2. A tel URI has the format tel:phonenumber, where phonenumber is the global or local phone number3.
• If a user of a SIP endpoint registered to an Expressway configured with only the cisco.com domain physically dials “6501”, the Expressway will assume that the user is dialing a SIP URI within the same domain, and append the domain name to the user part of the SIP URI4. Therefore, the alias presented in the SIP INVITE message will be 6501@cisco.com.
• If the user wants to dial a different domain, they must enter the full SIP URI, such as sip:6501@example.com4.
• If the user wants to dial a tel URI, they must prefix the phone number with a plus sign (+), such as tel:+65014.

References := 1: RFC 3261 - SIP: Session Initiation Protocol 2: RFC 3261 - SIP: Session Initiation Protocol 3: RFC 3966 - The tel URI for Telephone Numbers 4: Cisco Expressway Administrator Guide (X14.0) - Dial Plan Configuration

 The role of the conference alias in the Cisco Conductor is to be dialed by a user and Conductor uses that alias to determine the conference template. This is because the conference alias is a unique identifier for a conference that is configured on the Conductor. The conference alias can be a SIP URI, an H.323 ID, or an E.164 number. When a user dials the conference alias, the Conductor matches the alias to a conference template, which defines the settings and parameters for the conference, such as the bridge pool, the media resources, the layout, the encryption, and so on1.

The other options are not correct because:

• It determines the bridge pool to be selected for a conference: This is not true, as the bridge pool is determined by the conference template, not the conference alias. The bridge pool is a group of conference bridges that can host the conference. The conference template specifies which bridge pool to use for a conference, based on the priority, availability, and load balancing of the bridges2.
• The conference template determines the bridge pool to be selected for a conference: This is true, but it does not describe the role of the conference alias, which is the question. The conference template determines the bridge pool, but the conference alias determines the conference template1.
• It selects the Expressway to used for media traversal: This is not true, as the Expressway is not involved in the selection of the conference alias or the conference template. The Expressway is a network device that provides firewall traversal, NAT, and encryption services for video communications. The Expressway does not have the capability to select or configure conferences or conference aliases3.

References := 1: Cisco TelePresence Conductor with Cisco Unified Communications Manager Deployment Guide (XC4.0) - Configuring Conference Aliases [Cisco TelePresence Conductor] - Cisco 2: Cisco TelePresence Conductor with Cisco Unified Communications Manager Deployment Guide (XC4.0) - Configuring Conference Templates [Cisco TelePresence Conductor] - Cisco 3: Cisco Expressway Basic Configuration Deployment Guide (X12.6) - Introduction [Cisco Expressway Series] - Cisco

?

• Cisco Meeting Server is a software application that provides scalable, secure, and interoperable video, audio, and web conferencing on premises or in the cloud1.
• Cisco Meeting Server has several components that can be enabled to expand the functionality of the server, such as2:
• The component that is used for interworking between different communication protocols is the H.323 Gateway2. The H.323 Gateway allows Cisco Meeting Server to communicate with endpoints and servers that use different protocols, such as H.323, SIP, or WebRTC. The H.323 Gateway can also provide protocol conversion, media encryption, and bandwidth management for the interworked calls2.
• Therefore, the correct answer is D. H.323 gateway.

References := 1: Cisco Meeting Server - Overview 2: Cisco Meeting Server Deployment Guide (2.9)

= TMS permissions are assigned to groups, and users are added to the groups. This is because TMS uses a role-based access control model, where different groups have different levels of access and privileges to the TMS functions and resources. Users can belong to one or more groups, and inherit the permissions of those groups. TMS provides some predefined groups, such as Administrators, Booking, Users, and Read Only, and also allows creating custom groups with specific permissions1.

The other options are not correct because:

• TMS permissions are directly assigned to individual users: This is not true, as TMS does not allow assigning permissions to individual users. Users can only have permissions through the groups they belong to1.
• TMS permissions can be imported from the Expressway or Cisco Unified Communications Manager: This is not true, as TMS does not support importing permissions from other systems. TMS permissions are configured and managed within the TMS application1.
• TMS permissions are configured in Active Directory, which is then synched to TMS: This is partially true, as TMS can integrate with Active Directory to authenticate users and import user accounts and groups. However, the permissions are not configured in Active Directory, but in TMS. The Active Directory groups are mapped to the TMS groups, and the permissions are applied accordingly2.

References := 1: Cisco TelePresence Management Suite Administrator Guide (15.16) - User Administration [Cisco TelePresence Management Suite] - Cisco 2: Cisco TelePresence Management Suite Administrator Guide (15.16) - Active Directory Integration [Cisco TelePresence Management Suite] - Cisco