Pass the Cisco Field Engineer 500-710 Questions and answers with CertsForce

According to the Cisco documentation1, the Conductor uses the source IP addresses of the incoming calls to determine which subzone they belong to, and then applies the appropriate search rules to route them to the appropriate conference bridge. The conference IP addresses, location IP addresses, and management IP addresses are not used by the Conductor for this purpose. References: 1: Cisco Video Infrastructure Design Guide, Chapter 4: Cisco Conductor, https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/uc_system/design/guides/videodg/vidguide/conductor.html#pgfId-1060629

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

A cluster is a group of servers or nodes that work together to provide high availability, scalability, and load balancing for applications and services1.

Network latency is the time it takes for a packet of data to travel from one point to another on a network2. It is usually measured in milliseconds (ms) and can be affected by various factors, such as distance, bandwidth, congestion, and routing2.

The maximum network latency allowed between peers in a cluster depends on the type and configuration of the cluster, as well as the application and service requirements. Different cluster solutions may have different latency thresholds and recommendations for optimal performance and stability345.

one of the common cluster solutions that has a maximum network latency requirement is the Windows Server Failover Clustering (WSFC)3. WSFC is a feature of Windows Server that enables servers to work together to provide continuous service for applications and services, such as SQL Server, Exchange Server, and Hyper-V3.

According to the Microsoft documentation, the maximum network latency allowed between peers in a WSFC cluster is 100 ms one way3. This means that the round-trip latency between any two nodes in the cluster should not exceed 200 ms. This requirement applies to both the private and public network connections between cluster nodes, and is based on the assumption that the network is reliable and stable3.

The other options are incorrect because:

A. 30 ms total is too low for a realistic network latency requirement between peers in a cluster. It is unlikely that any network can consistently provide such a low latency across different locations and conditions2.

B. 300 ms total is too high for a network latency requirement between peers in a cluster. It is likely that such a high latency would cause performance degradation, timeouts, and failures for applications and services that rely on the cluster2.

D. 150 ms one-way is also too high for a network latency requirement between peers in a cluster. It would imply a round-trip latency of 300 ms, which is above the threshold for WSFC and other cluster solutions345.

References := 1: What is a Cluster? 2: What is Network Latency? 3: Windows clustering and geographically separate sites 4: Cluster Network Latency - Troubleshooting Guide 5: Support Policies for RHEL High Availability Clusters - Cluster …

To add individual endpoints or infrastructure to the TMS database, an engineer needs to use the System > Navigator menu in the TMS application. The System > Navigator menu allows the engineer to browse, search, and manage systems in the TMS database. The engineer can add systems manually, by IP address, by DNS name, or by discovery12. The engineer can also edit, delete, move, or group systems in the System > Navigator menu12. References := 1

https://community.cisco.com/t5/telepresence-and-video-infrastructure/add-endpoints-to-tms-hosted-solution/td-p/1998838

https://community.cisco.com/t5/telepresence-and-video-infrastructure/add-cucm-registered-endpoint-to-tms/td-p/2143976

 = Cisco WebEx Telepresence is a cloud-based service that enables on-premises endpoints to register to Cisco infrastructure in the cloud and join video calls with other WebEx Telepresence users or compatible standards-based conferencing systems over the Internet. It also provides features such as directory, presence, scheduling, and management for the registered endpoints. Cisco WebEx Telepresence works with various Cisco TelePresence endpoints, such as multipurpose, integrator, personal, and video mobility systems123. References := 123

To build a TMS phone book, an engineer needs to follow these steps:

Create a source: A source is a collection of contacts that can be imported from various sources, such as Cisco TMS endpoints, manual lists, Active Directory, H.350 directories, file-based phone books, gatekeepers, or other TMS phone books. A source can be created by going to Phone Books > Manage Phone Book Sources and clicking on New12.

Create a phone book: A phone book is a logical grouping of contacts that can be assigned to one or more systems. A phone book can be created by going to Phone Books > Manage Phone Books and clicking on New12.

Connect the source to the phone book: A source can be connected to a phone book by selecting the phone book and clicking on Connect. Then, the engineer can choose the source from the list and click on OK12.

Set the phone book on systems: A phone book can be set on systems by selecting the phone book and clicking on Set on Systems. Then, the engineer can choose the systems from the list and click on OK12. References :=https://www.cisco.com/c/en/us/support/docs/conferencing/telepresence-management-suite-tms/215223-tms-phone-books-troubleshoot-guide.html

https://www.cisco.com/c/en/us/support/docs/conferencing/telepresence-management-suite-tms/211266-How-to-Create-a-File-Based-Phonebook-in.html

An endpoint registration request message that is sent to a SIP Server contains the following two options1:

E.164 address: This is the telephone number of the endpoint, which is used to identify the endpoint in the SIP domain. The E.164 address is also known as the Address of Record (AOR), which is a high-level name that is assigned to a SIP entity without regard to the device or devices that name might use. The format for an AOR is sip:user@domain (e.g. 2025551234@sip.plivo.com)2.

SIP URI: This is the location of the endpoint, which is identified by an IP address, port number and protocol. The SIP URI is used to reach the endpoint for establishing a media session. Theformat for a SIP URI is sip:user@host:port;transport (e.g. sip:2025551234@192.168.1.120:5060;udp)1.

The other options are not correct because:

SIP Server IP address: This is not included in the endpoint registration request message, as the endpoint already knows the IP address of the SIP Server to which it is sending the message. The SIP Server IP address is usually configured in the endpoint settings as the Registration Domain or the Outbound Proxy3.

Endpoint IP address: This is not included in the endpoint registration request message, as it is already part of the SIP URI. The endpoint IP address is the host part of the SIP URI, which indicates the location of the endpoint1.

E.164URI: This is not a valid option, as there is no such thing as an E.164URI. The E.164 address is not a URI, but a telephone number that is used to identify the endpoint in the SIP domain2.

References := 1: Cisco Unified Communications System 9.0 SRND - SIP Infrastructure 2: SIP signalling- the registration process and setting up a SIP call 3: Programmable Voice - SIP Registration | Twilio

The Expressway supports two protocols for traversing media across a firewall: H.460.18/19 and Assent. Both protocols use a TCP/TLS connection established from the client to the server for signaling, and UDP ports for media. The difference is that H.460.18/19 is a standard protocol for H.323 firewall traversal, while Assent is a Cisco proprietary protocol for SIP firewall traversal123

H.460.18/19 is an extension of the H.323 protocol that allows endpoints to communicate with each other through a firewall or a NAT device. H.460.18 defines the signaling procedures for establishing a TCP/TLS connection between the endpoint and the Expressway-E, which acts as the traversal server. H.460.19 defines the media procedures for sending and receiving RTP/RTCP packets through the same TCP/TLS connection orthrough UDP ports. The Expressway-E uses media demultiplexing to route the media packets to the correct destination124

Assent is a protocol that is similar to H.460.18/19, but it is used for SIP firewall traversal of media. The signaling is traversed through a TCP/TLS connection established from the client to the server. The media is traversed through UDP ports that are negotiated using the SDP offer/answer exchange. The Expressway-E uses media demultiplexing to route the media packets to the correct destination123

References := 1Cisco Expressway Administrator Guide (X14.0) - Firewall Traversal 2Cisco Expressway IP Port Usage for Firewall Traversal 3Firewall Traversal and NAT Traversal On Cisco Expressway Explained With Wireshark 4Firewall Traversal Mechanisms > CCIE Collaboration Quick Reference

Branding on the Cisco Meeting Server refers to the customization of prompts and backgrounds for the web app, SIP calls, and IVR messages. With a single branding license, you can customize the following aspects of the Cisco Meeting Server1:

The sign-in page background image, icon, and logo for the web app

The text strings for the web app

The audio messages and background image for the SIP calls

The audio messages and background image for the IVR messages

The invitation text for the web app

You can apply branding to the Cisco Meeting Server in two ways: web server hosted branding or locally hosted branding1. Web server hosted branding requires a web server that hosts the branding files and provides the URLs to the Cisco Meeting Server API. Locally hosted branding requires uploading the branding files to the Cisco Meeting Server using the API1.

With web server hosted branding, you can use a single branding license to customize multiple logos and multiple backgrounds on multiple web bridges. This means that you can have different branding for different web bridges, depending on the web bridge profile that you associate with the web bridge1. For example, you can have a web bridge with a blue logo and background for one web bridge profile, and another web bridge with a red logo and background for another web bridge profile.

With locally hosted branding, you can use a single branding license to customize a single logo and single background on a single web bridge. This means that you can only have one branding for one web bridge, regardless of the web bridge profile that you associate with the web bridge1. For example, you can only have a web bridge with a green logo and background for any web bridge profile.

Therefore, the correct answer is B and E, as they describe the capabilities of web server hosted branding and locally hosted branding, respectively. References := 1: Cisco Meeting Server 3.5 Customization Guidelines,

= Transforms are a way of modifying the destination alias or address of a call before it is routed to the appropriate zone or service. Transforms can be applied at different stages of the call routing process, such as pre-search, search rules, or call policy. The Cisco recommended method to add a domain to a numeric E.164 alias dialed by an H.323 endpoint is to use a pre-search transform. A pre-search transform is applied to the destination alias before any searches are sent to external zones. This allows the Expressway to standardize the destination aliases originating from both H.323 and SIP devices. For example, if the called address is an H.323 E.164 alias “01234”, the Expressway can automatically append the configured domain name (in this case example.com) to the called address (that is, 01234@example.com making it into a URI), before attempting to set up the call. This way, the Expressway can route the call to the correct destination zone or service based on the full URI. References := Cisco Expressway-E and Expressway-C Basic Configuration Deployment Guide (X14.0) - Routing ConfigurationCisco Expressway Administrator Guide (X14.0) - Dial Plan and Call Processing