You are configuring digest authentication so that the identity of SIP phones can be challenged by the UCM to
which they are connected. After configuring an appropriate security profile, you apply the profile to each SIP
phone on the network. After creating a digest user in the UCM Administration End User window, you notice that
a Cisco 7961G IP phone is not able to authenticate with UCM.
Which of the following should you do? (Select 2 choices.)
A.
Associate the digest user with the SIP phone in UCM Administration.
B.
Configure the SIP realm on a SIP trunk.
C.
Reset the phone.
D.
Specify digest credentials in the Application User Configuration window.
E.
Upload the configuration file to the TFTP server.
Explanation:
You should associate the digest user with the Session Initiation Protocol (SIP) phone in Cisco Unified
Communications Manager (UCM) Administration and then reset the Cisco 7961G IP phone in order to enable
the phone to use digest authentication to verify its identity with the UCM to which it is connected.The digest credentials for most Cisco IP phones are stored in the phone’s configuration file, which is
downloaded from a Trivial File Transfer Protocol (TFTP) server when the phone is started or reset. On Cisco
7940G and 7960G SIP IP phones, the digest credentials must be manually entered from the IP phone.
Digest credentials consist of a unique user ID, password, and digest realm. UCM generates a Message Digest
5 (MD5) hash from these values and a random number. A checksum is generated from the hash. The user
name and checksum are then stored in the UCM database in an encrypted format.
To enable UCM to authenticate a SIP phone, you should first configure a security profile for SIP phones and
verify that the Enable Digest Authentication check box has been selected. Next, you should apply the security
profile to the SIP phones that you want to be authenticated. After the security profile has been created and
applied, you should configure a digest user in the UCM Administration End User window, where you specify the
digest user ID and password that you want the SIP phone to use to authenticate. Finally, you must associate
the digest user with the SIP phone that you want to be authenticated and reset that SIP phone so that it
downloads its new configuration. The new configuration contains the digest credentials.
You do not need to upload the SIP phone configuration file to the TFTP server. UCM updates the configuration
file so that it can be downloaded from the TFTP server by the IP phones. However, for security reasons, you
might want to ensure that TFTP traffic between the server and the IP phones is encrypted. Otherwise, the
digest credentials will be included in a configuration file that is sent across the network as clear text.
You do not need to specify digest credentials in the Application User Configurationwindow. The Application
User Configuration window can be used to specify digest credentials for SIP applications that you want to
authenticate with UCM.
There is nothing in this scenario to indicate that you should configure the SIP realm on a SIP trunk. You would
need to configure a SIP realm if you were receiving digest authentication challenges over a SIP trunk.https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/9_0_1/secugd/
CUCM_BK_CCB00C40_00_cucm-security-guide-90/CUCM_BK_CCB00C40_00_cucm-securityguide_chapter_01100.html#CUCM_TK_S2044B79_00
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/9_0_1/secugd/
CUCM_BK_CCB00C40_00_cucm-security-guide-90/CUCM_BK_CCB00C40_00_cucm-securityguide_chapter_01.html#CUCM_RF_D4C84CE2_00
More new 210-060 Exam Questions:
QUESTION 109
The IP phone of user A is registered with Cisco Unified Communications Manager subscriber1 while the IP phone of user B is registered with subscriber2. User A is speaking to user B on an active call. A junior network engineer mistakenly reboots subscriber1.
What effect does this have on the call?
A. User A can hear B, but B cannot hear A.
B. User A cannot hear B, but B can hear A.
C. This action drops the call.
D. This action does not affect the call.
E. The call remains active, but quality may suffer.
Answer: D
QUESTION 110
A company has invested in an on-premises VoIP solution. For design simplicity purposes, network engineers have opted to use a single VLAN for both data and voice traffic. Shortly after implementing IP phones, customers are reporting problems.
Which two potential problems may be reported about the calls as a result of this design decision? (Choose two.)
A. jitter detected in voice calls
B. problems with email latency
C. slow internet download speeds
D. choppy voice calls
E. PCs are getting IP addresses but phones are not
Answer: AD
QUESTION 111
A network engineer receives a report about poor quality on an active call between the IP phone of user A over the WAN to the IP phone of user B. Using web access to the phone, the network engineer remotely checks call statistics such as jitter, network delay, and packet loss. Calculated packet loss is 3%, average jitter is 20 ms, network delay is 1 ms, and conceal seconds is 7.
What is the most likely problem with this call?
A. Calculated packet loss is too high.
B. Average jitter is too high.
C. Network delay is too high.
D. Conceal seconds are too high.
Answer: A
QUESTION 112
Which command allows the telephony service of a Cisco Unified Communications Manager Express router to be associated to loopback address 192.168.143.44?
A. !
telephony-service
max-ephones 4
max-dn 8
ip source-address 192.168.143.44
!
B. !
telephony-service
max-ephones 4
max-dn 8
ip bind src-addr 192.168.143.44
!
C. !
telephony-service
max-ephones 4
max-dn 8
source-address ipv4:192.168.143.44
!
D. !
telephony-service
max-ephones 4
max-dn 8
ip address 192.168.143.44
!
Answer: A
QUESTION 113
Which four devices can be used to provide analog ports, traditional phones, and fax machines? (Choose four.)
A. Cisco VG224 Analog Voice Gateway
B. Foreign Exchange Station Voice Interface Card
C. Cisco High Density VoiceFax Network Module
D. Cisco ATA190 Analog Telephone Adapter
E. Cisco VG350 Analog Voice Gateway
F. Cisco Unified Border Element
G. Foreign Exchange Office Voice Interface
Answer: ABDE
QUESTION 114
Client A in X site uses an IP phone to call client B in Y site. Engineers have selected SCCP as the default VoIP signaling protocol. Which network path will the call signaling take when client A calls client B?
A. IP phone X > CUCM subscriber > IP phone Y using TCP port 1000
B. IP phone X > CUCM subscriber > IP phone Y using TCP port 2000
C. IP phone X > CUCM subscriber > IP phone Y using UDP port 1000
D. IP phone X > IP phone Y using TCP port 2000
E. IP phone X > IP phone Y using TCP port 1000
Answer: B
QUESTION 115
Which Cisco Unified Communications Manager protocol communicates with collaboration endpoints?
A. SCCP
B. RTP
C. SIP
D. CDP
Answer: C
QUESTION 116
Which two tools can be used to measure the quality of a VoIP call? (Choose two.)
A. QoS configuration tool
B. mean opinion score tool
C. bulk administration tool
D. jitter compensation tool
E. rFactor tool
Answer: BE
QUESTION 117
What is the recommended maximum one-way latency for voice and video networks?
A. 100 ms
B. 150 ms
C. 200 ms
D. 300 ms
Answer: B
QUESTION 118
An engineer must configure QoS for a VoIP network. Which option is the best QoS model for this task?
A. IntServ
B. DiffServ
C. MPLS
D. FIFO
Answer: B
QUESTION 119
User A notices echo on a call with user B. Both users are using Cisco VoIP phones. User B is using a headset, and user A is using a handset. What is the most likely source of the echo?
A. user A handset
B. user B headset
C. disabled echo cancellation on user A phone profile
D. disabled echo cancellation on user B phone profile
Answer: B
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