單項(xiàng)選擇題A GRE tunnel is configured between a local and a remote site. Where should the service policy be applied to classify packets based on the pretunnel header? ()

A.in global configuration mode, apply the service policy and use the qos pre-classify
B.in global configuration mode, apply the service policy but do not use the qos pre-classify command
C.apply the service policy on the physical interface but do not use the qos pre-classify
D.apply the service policy on the tunnel interface but do not use the qos pre-classify command 
E.apply the service policy on the tunnel interface and use the qos pre-classify command


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1.多項(xiàng)選擇題What are two of the three predefined classes for the Cisco SDM wizard?()

A.business-critical 
B.high priority 
C.middle class
D.voice
E.best effort
F.scavenger

2.多項(xiàng)選擇題Which two statements are true about the application of QoS in a converged network?()

A.end-to-end network delay times that exceed 50 ms for real-time traffic are considered unacceptable
B.end-to-end network delay times that exceed 250 ms for real-time traffic are considered unacceptable
C.end-to-end network delay is not a factor as long as the  delay is consistent
D.some packet loss can be corrected by codec algorithms
E.RSVP handles voice packet retransmission
F.fragmentation is a result of packet loss

3.單項(xiàng)選擇題What best describes an FXO interface?()

A.analog trunks that provide the Survivable Remote Site Telephony (SRST) feature 
B.analog trunks that provide VoIP gateway functionality 
C.analog trunks that connect a gateway to plain old telephone service (POTS) device such as analog phones, fax machines, and legacy voice-mail systems  
D.analog trunks that connect a gateway to a central office (CO) or private branch exchange (PBX)

5.單項(xiàng)選擇題Which statement is true about a distributed call control environment and the processing of dialed digits from an IP phone?()

A.The router that is directly connected to the IP phone will look up the called number in its call routing table.
B.The router that is directly connected to the IP phone will inform its call agent when a service request is detected.
C.When the IP phone is picked up,the IP phone initiates the service request.
D.The directly connected router passes the collected digits to its call agent,and the call agent looks up in its call-routing table the called number.

6.多項(xiàng)選擇題What are two major sources of delay that can be managed by QoS in voice-enabled networks?()

A.propagation delay
B.voice packet serialization delay
C.congested egress queues
D.header overhead
E.packets dropped because of CRC errors

7.多項(xiàng)選擇題Which two statements are true about voice ports on a router?()

A.Calls to the PSTN can be made via the use of FXO or T1/E1 trunk ports.
B.Calls to the PSTN can be made via FXS or T1/E1 trunk ports.
C.Analog and IP phones can be connected to the VoIP network via FXO or T1/E1 trunk ports.
D.Calls between analog phones that are attached to the FXS ports in a VoIP network can be completely processed by voice-enabled routers.

8.單項(xiàng)選擇題Which statement is true about Foreign Exchange Station (FXS) ports on a router?()

A.The FXS interface allows an analog connection to be directed at the public switched telephone network (PSTN’s) central office.
B.The FXS interface connects directly to a standard telephone,fax machine,or similar device and supplies ring,voltage,and dial tone.
C.The FXS interface connects directly to an IP phone and supplies ring,voltage,and dial tone.
D.The FXS interface connects directly to ISDN voice channels.

9.多項(xiàng)選擇題Which two statements are true about the digital audio in a VoIP network?()

A.Standard encoding techniques create an uncompressed digital data rate of 4000 bps.
B.Standard encoding techniques create an uncompressed digital data rate of 8000 bps.
C.Standard encoding techniques create an uncompressed digital data rate of 64,000 bps.
D.Voice quality is not a concern if compression is not used.
E.Two methods of quantization are linear and logarithmic.
F.Two methods of compression are u-law and a-law.

10.多項(xiàng)選擇題Which two statements are true about analog to digital conversion of voice signals for use in digital telephony networks?()

A.The three required steps in the analog to digital conversion are sampling,encoding,and compression.
B.The three required steps in the analog to digital conversion are sampling,quantization,and compression.
C.The three required steps in the analog to digital conversion are sampling,quantization,and encoding.
D.The output of the sampling process is a pulse code modulation (PCM) signal.
E.The output of the sampling process is a pulse amplitude modulation (PAM) signal.

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