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Exam 500-710 Review - New 500-710 Dumps Questions

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Cisco 500-710 Certification Exam is designed for professionals who want to demonstrate their expertise in implementing and configuring video infrastructure solutions using Cisco technologies. 500-710 exam covers a broad range of topics related to video infrastructure implementation, including video conferencing, video collaboration, and video networking. Cisco Video Infrastructure Implementation certification is aimed at individuals who work in a variety of roles, including video engineers, network administrators, and technical support specialists.

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New 500-710 Dumps Questions - 500-710 Latest Questions

As you know, there are so many users of our 500-710 guide questions. If we accidentally miss your question, please contact us again and we will keep in touch with you. Although our staff has to deal with many things every day, it will never neglect any user. With the development of our 500-710 Exam Materials, the market has become bigger and bigger. Paying attention to customers is a big reason. And we believe that with the supports of our worthy customers, our 500-710 study braindumps will become better.

Cisco 500-710 Exam is structured in such a way that it covers a wide range of topics in the implementation of video infrastructure products. These topics include implementing Cisco TelePresence Server, Cisco TelePresence Management Suite, Cisco TelePresence Content Server, and Cisco TelePresence MCU. All these technologies are essential for the configuration of video infrastructure products and their maintenance and troubleshooting.

Cisco Video Infrastructure Implementation Sample Questions (Q61-Q66):

NEW QUESTION # 61
Which method does the Conductor use to determine the source of an incoming call to a rendezvous type conference?

A. management IP addresses
B. location IP addresses
C. source IP addresses
D. conference IP addresses

Answer: B

NEW QUESTION # 62
Which two statements about the relationship between the Expressway series and the Cisco Unified Communications Manager are true? (Choose two.)

A. The Expressway series only supports SIP-based communications.
B. The Expressway series can never operate as a standalone call-control solution.
C. The Expressway solution can be peered with a Cisco Unified Communications Manager to allow for greater capabilities.
D. The Expressway series may operate as a standalone call-control solution.
E. The Expressway series requires the peering with a Cisco Unified Communications Manager to act as a call-control device.

Answer: A,D

NEW QUESTION # 63
If a user of a SIPendpoint registered to an Expressway configure with only thecisco.comdomain physically dials "6501", which alias is presented in the SIP INVITE message?

A. 6501@<ip address of the Expressway>
B. 6501@<ip address of the endpoint>
C. 0
D. 6501@cisco.com

Answer: D

Explanation:
* 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

NEW QUESTION # 64
Which two options are included in an endpoint registration request message that is sent to a SIP Server?
(Choose two.)

A. E.164URI
B. SIP URI
C. Endpoint IP address
D. SIP Server IP address
E. E.164 address

Answer: B,E

Explanation:
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

NEW QUESTION # 65
Cisco Meeting Server uses four API commands to interact with the server. Which API command is used to modify a configuration setting on the Cisco Meeting Server?

A. GET
B. PUT
C. DELETE
D. POST

Answer: B

Explanation:
= The Cisco Meeting Server uses a RESTful API to interact with the server. The API commands are sent via HTTP requests to the web admin service, which processes them and returns the response. The four basic API commands are GET, POST, PUT, and DELETE. Each command has a different function and syntax, as explained below12:
* GET: Retrieves existing information from the server. The syntax is GET /api/v1/<object>
/<id> where <object> is the type of object to retrieve, such as call, participant, space, etc., and <id> is the unique identifier of the object, such as a UUID or a name. For example, GET /api/v1/calls
/12345678-1234-1234-1234-123456789012 will return the information about the call with the specified UUID.
* POST: Creates new information on the server. The syntax is POST /api/v1/<object> where <object> is the type of object to create, such as call, participant, space, etc. The request body contains the attributes and values of the object to create, in JSON format. For example, POST /api/v1/spaces with the request body {"name": "Test Space", "uri": "test@domain.com"} will create a new space with the specified name and URI.
* PUT: Modifies existing information on the server. The syntax is PUT /api/v1/<object>
/<id> where <object> is the type of object to modify, such as call, participant, space, etc., and <id> is the unique identifier of the object, such as a UUID or a name. The request body contains the attributes and values of the object to modify, in JSON format. For example, PUT /api/v1/spaces/12345678-1234-
1234-1234-123456789012 with the request body {"name": "New Name"} will change the name of the space with the specified UUID to "New Name".
* DELETE: Deletes existing information from the server. The syntax is DELETE /api/v1/<object>
/<id> where <object> is the type of object to delete, such as call, participant, space, etc., and <id> is the unique identifier of the object, such as a UUID or a name. For example, DELETE /api/v1/spaces
/12345678-1234-1234-1234-123456789012 will delete the space with the specified UUID.
In this question, the API command that is used to modify a configuration setting on the Cisco Meeting Server is PUT, as it allows changing the attributes and values of an existing object on the server.
References := Cisco Meeting Server Basic API Functions, Cisco Meeting Server - Programming Guides

NEW QUESTION # 66
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