Online Access Free 642-437 Exam Questions

DRAG DROP
Click and drag the feature on the left to the category it belongs to on the right.

Correct Answer:

Explanation:
Gateway: Supports Analog Faxes and Modems on a Voip Network
Performs Call Setup and teardown between Voip Networks & the PSTN.
CUBE. Interconnects segments of the same or different VoIP networks using different media types.
Interconnects segments of the same or different VoIP networks using different media types.
Gateway Functionality: Gateways are responsible Media stream handling and speech path
integrity, DTMF relay, Fax relay and pass-through, Digit translation and call processing, Dial peers
and codec filtering, Carrier ID handling, Termination and re-origination of signaling and media.
The Cisco Unified Border Element is a session border controller designed to provide easy, secure,
and cost-effective connectivity between independent unified communications networks or network
domains for different enterprises. It provides interconnection between incompatible applications
within the enterprise network, between different enterprises for business-to-business applications,
and between enterprise networks and service provider Session Initiation Protocol (SIP) trunks.
The Cisco Unified Border Element provides key session management capabilities, H.323 and SIP
interworking functions, and network-to-network interface security and demarcation capabilities. It
performs most of the same functions of a public switched telephone network (PSTN)-to-IP
gateway but joins two VoIP call legs. Media packets can either flow through (thus hiding the
networks from each other) or around the Cisco Unified Border Element platform.
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-overview_ps10591_TSD_Products_Configuration_Guide_Chapter.html

What is the total bandwidth per call in a link, if you use a G.729 (high-complexity codec) with a 40byte sample and Frame Relay with cRTP?

• A.19,200
• B.11,200
• C.9,600
• D.7,420

A small office needs to provide outbound dialing and in-bound DID without the cost of a T1 circuit. All signaling is loop start. Which analog port configuration will support these requirements?

• A.voice-port 0/0/0 description fxs-did signal did loop-start ! voice-port 0/1/0 description fxo signal loop-start ! dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0/0/0 ! dial-peer voice 90 pots destination-pattern 9T port 0/1/0
• B.voice-port 0/1/0 signal did loop-start ! dial-peer voice 1 pots incoming called-number . ! dial-peer voice 90 pots destination-pattern 9T port 0/1/0
• C.voice-port 0/0/0 signal did loop-start ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! dial-peer voice 90 pots destination-pattern 9T port 0/0/0
• D.voice-port 0/0/0 signal loop-start ! voice-port 0/1/0 signal loop-start ! dial-peer voice 1 pots incoming called-number T direct-inward-dial ! dial-peer voice 90 pots destination-pattern 9T port 0/1/0

Refer to the exhibit.

When 5551234 is being matched with the outgoing dial peer that is shown in the exhibit, which of the following called numbers will be sent to the VoIP network?

• A.555
• B.1234
• C.5
• D.51234
• E.5551234
• F.Null

Voice packets are arriving at a destination with a variance of between 20 and 50 milliseconds. If the jitter buffer has a capacity of 30 milliseconds, what is the impact on the audio at the receivers IP phone? Voice packets are arriving at a destination with a variance of between 20 and 50 milliseconds. If the jitter buffer has a capacity of 30 milliseconds, what is the impact on the audio at the receivers IP phone?

• A.There will be no impact the audio stream because the audio packets are arriving in the jitter buffer window.
• B.The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 milliseconds to avoid speech gaps.
• C.The DSP will automatically increase the jitter buffer size after sampling the range of incoming voice packets to accommodate the wider range in variation of voice packet arrival times to avoid voice gaps.
• D.The IP phone will negotiate in mid-call a lower bandwidth codec to reduce the delay in the arrival of voice packets to avoid voice gaps.