Which two VoIP capabilities are utilized when the CallManager and the gateway are configured for Survivable Remote Site Telephony？（）

A . cs3 B . cs7 C . ef D . af41 E . af31 F . 0（default）

A . MCU B . Application server C . Gateway D . Gatekeeper

A . TCP retransmits B . UDP priority C . Bursty D . Delay sensitive E . Greedy F . Drop sensitive

A . Delay should be no more than 10 ms. B . Loss should be no more than1 percent. C . Jitter should be less then 40 ms. D . Managed bandwidth is strongly recommended for voice control traffic.

A .  Type 1 B .  Type 2 C .  FXS D .  FXO E .  E&M

A . allowed codecs B . native VLAN C . auxiliary VLAN D . Cisco Unified Communications Manager IP address E . RSTP

A . Use WFQ， CBWFO， and LLO to prioritize delay-sensitive packets B . Use WFO， CBWFO， and LLO to compress the payload. C . Give TCP packets priority over UDP packets. D . Use control plane policing for cRTP queuing. E . Use stacker and predictor to compress the payload.

A . The use of a queuing method that will give VoIP traffic strict priority over other traffic B . The use of RTP header compression for the VoIP traffic C . The proper classification and marking of the traffic as close to the source as possible D . The use of 802.1QinQ trunking for VoIP traffic E . The use of WRED

A . Trust boundaries are configured in the core of the network to provide the most efficient  forwarding based upon QoS markings， B . Setting trust boundaries at the edge of the network allows intermediate hop devices to  administer QoS policies without detailed packet identification. C . Trust boundaries are used to determine which QoS mechanism will be applied to the traffic.   D . At the trust boundaries， the untrusted traffic will be marked with a new QoS value appropriate  for the policy in place at the point where the traffic entered the campus network.   E . Trust boundaries define the firewall rules for QoS admission into a network.

A . The use of a queuing method that will give VoIP traffic strict priority over other traffic   B . The use of RTP header compression for the VoIP traffic. C . The proper classification and marking of the traffic as close to the source as possible D . The use of 802.1QinQ trunking for VoIP traffic E . The use of WRED for the VoIP traffic

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 packetization size B . the serialization of the interface C . the quantization D . the packet rate E . the UDP overhead F . the TCP overhead

A . serialization delay B . overrun C . full buffers and queues D . propagation delay E . shaping delay F . TCP header error

A .  Packets can arrive out of order. B .  Packets must take the same path to the destination. C .  Once packets are received at the destination router， the router performs sampling to ensure the quality of the voice traffic. D .  Packets can experience varying delays across the network. E .  Packets containing voice samples are sequentially numbered by default， allowing for reordering upon arrival at the terminal router along the path.

A .  CAC is used to allocate enough bandwidth for the number of calls that are allowed on the VoIP network. B .  CAC is used to protect the quality of the voice calls by preventing call establishment if enough resources are not available on the network. C .  CAC provides queuing mechanisms for voice calls if the packets exceed the configured or budgeted rate. D .  In a VoIP network the CAC can be replaced with the LLQ QoS mechanism if it has been properly implemented.

A .  Delay should be no more than 10 ms. B .  Loss should be no more than 1 percent. C .  Jitter should be less then 40 ms. D .  Managed bandwidth is strongly recommended for voice control traffic.

A .  voice stream mixing for conference calling B .  voice gateway C .  call admission control D .  analog or digital phone interfaces E .  XML application services F .  large scale dial plan applicatio

A .  Voice carrier stream utilizes Real-Time Transport Protocol (RTP) to carry the audio/media  portion of the VoIP communication.   B .  Voice traffic data flow involves large volumes of large packets.   C .  Because a packet loss involves a small amount of data, voice traffic is less affected by packet  losses than traditional data traffic is.   D .  Voice packets are encapsulated in TCP segments to allow for proper sequencing during  delivery.  E .  Voice packets are very sensitive to delay and jitter.&e

A .  Use WFQ， CBWFQ， and LLQ to prioritize delay-sensitive packets. B .  Use WFQ， CBWFQ， and LLQ to compress the payload. C .  Give TCP packets priority over UDP packets. D .  Use control plane policing for cRTP queuing. E .  Use stacker and predictor to compress the payload.

A . the VoIP endpoints have the intelligence to set up and control calls B . Call setup and control functionality is centralized in one call agent or cluster C . Call setup and control resides in call agents that are distributed throughout the network D . Each VoIP device has separate call control， voice packetization， and transport mechanism

A . Cisco Proprietary payload compression B . Predictor payload compression C . RTP header compression D . TCP header compressio

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 gateway voice-capable device is configured to support call control directly performing call setup， call maintenance， and call teardown B . The gateway voice-capable device has a local call routing table configured through a routing protocol. C . The gateway voice-capable device relies on a call agent to support call control. D . The gateway voice-capable device relies on a gatekeeper to support call setup， call maintenance， and call teardown.

A . gatekeeper B . Call Routing C . Call Admission Control D . Digital Signal Processor