A.20 bytes
B.60 bytes
C.120 bytes
D.160 bytes

A.minimum jitter
B.maximum jitter
C.upstream standard packet deviation
D.average MOS value
E.average conceal rate

A.The phone does a directed call hangup on 802.11b，then does an auto-redial on 802.11a，creating a  new call.
B.The phone does a directed call hangup on 802.11b， then the user does a redial on 802.11a，creating a  new call.
C.The phone does a disassociation on 802.11b and an association on 802.11a，and call remains up if the  subnet has not changed.
D.The phone does not roam from 802.11b to 802.11a when on an active call.
E.The phone must switch network profiles to roam from 802.11b to 802.11a.

A.RSSI of the client at the access point
B.WMM state
C.current MOS value
D.current RF channel
E.current transmit data rate

A.disable higher data rates
B.set DTIM = 1
C.disable RRM
D.disable broadcast SSID
E.disable multicast

A.duplicate sender packets
B.unicast packet retry rate
C.unicast packet retry ratio
D.association rejections

A.user request using the keypad of the phone and HTTP download
B.IT department scheduled upgrade using HTTP download
C.TFTP download based on version control through Cisco Unified Communications Manager
D.FTP download based on version control through Cisco Unified Communications Manager
E.HTTP download based on version control through Cisco Unified Communications Manager

A.Layer 3 with Layer 3 Security set to None and Web Policy set to Authentication
B.Layer 3 with Layer 3 Security set to None and Web Policy set to VPN Pass-Through
C.Layer 3 with Layer 3 Security set to VPN Pass-Through
D.Layer 2 with Layer 2 Security set to 802.1X

A.Yes，by modifying the LWAPP frame header DSCP value in transit.
B.Yes，by modifying the LWAPP frame header 802.1p value in transit.
C.No，the WMM queue classification parameter is transported inside the LWAPP tunnel.
D.No，the WMM queue classification parameter is determined at the AP based on the original client IP  header DSCP value.

A.Voice traffic should be marked to DSCP EF per the QoS baseline and RFC3246.
B.Voice traffic loss should be no more than 3%.
C.One-way latency （mouth-to-ear） should be less than 150 msec.
D.Average one way jitter should be targeted to less than 30 msec.
E.21-320 Kbps of guaranteed priority bandwidth is required per call depending on sample rate，codec，and  layer 2 overhead.