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

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)

A.sampling
B.packetization
C.quantization
D.serialization

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.

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

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.

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.

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.

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.

A.Cisco AutoQoS can be used repeatedly to apply a single QoS policy to multiple interfaces.
B.Cisco AutoQoS includes an optional web-based GUI for automating the configuration of QoS services.
C.Cisco AutoQoS provides capabilities to automate VoIP deployments.
D.The auto qos global configuration command is used to configure Cisco AutoQoS.
E.The Modular QoS CLI （MQC） is the best way to fine tune QoS configurations.
F.The SDM QoS wizard is the fastest way to implement QoS.