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Cisco 642-437 Actual Tests : Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0)
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Cisco 642-437 Exam Syllabus Topics:
| Section | Objectives |
|---|---|
| Topic 1: Cisco Unified Border Element (CUBE) | - CUBE features and configuration
|
| Topic 2: Troubleshooting | - Troubleshooting voice and QoS issues
|
| Topic 3: Quality of Service (QoS) | - QoS mechanisms
|
| Topic 4: Voice Gateways and Call Routing | - Gateway implementation and configuration
|
| Topic 5: Cisco Unified Communications Manager Express | - CUCME configuration
|
| Topic 6: VoIP Fundamentals | - Voice over IP basics
|
Cisco Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Sample Questions:
Which command should you use to configure a dial-peer to support T.38 fax relay and to use Cisco fax relay if T.38 fails?
- A. fax protocol t38 fallback cisco
- B. fax relay t38 cisco
- C. fax t38 fallback cisco
- D. fax relay t38 backup cisco relay
Correct Answer: A 🗳️
What is the maximum percentage of packet loss that can be accepted on VoIP traffic?
- A. 0
- B. less than 1%
- C. less than 2%
- D. less than 5%
Correct Answer: B 🗳️
Which two options are Cisco recommendations for enabling Cisco Unified Border Element to perform a CAC? (Choose two.)
- A. two Cisco Unified Border Element gateways to enable RSVP-based CAC, with the media flow-through command configured
- B. two Cisco Unified Communications Manager clusters interconnected using a Cisco Unified Border Element
- C. Cisco Unified Border Element gateways activating RSVP-based CAC, using the media flow-around command
- D. an interconnection of a Cisco Unified Border Element and a Cisco Unified Communications Manager to support both voice and video calls
- E. a voice service VoIP that is configured on the unique Cisco Unified Border Element to enable the CAC service
Correct Answer: A,B 🗳️
What should you use to configure endpoint addressing if you are using Cisco Unified Communications Manager Express with SRST?
- A. ephone-dn
- B. POTS
- C. dial-peer
- D. VoIP
Correct Answer: A 🗳️
DRAG DROP
Drag the functions that are associated with H.245 from the list on the left to the boxes on the right.
Correct Answer:
Explanation:
H.225 is responsible only for setting up the call and routing it to the proper destination.
H.225 does not have any mechanism for exchanging capabilities or setting up and tearing down media streams.
The called H.323 device is responsible for sending the IP address and port number that are used to establish the TCP connections for H.245 signaling.
This information can be sent by the called device in either the Alerting or Connect message.
When the originating H.323 device receives the IP address and port number for H.245 negotiations, it initiates a second TCP connection to carry out the necessary capabilities exchange and logical channel negotiations. This TCP session is primarily used to do four things: Master/slave determination-This is used to resolve conflicts that might exist when two endpoints in a call request the same thing, but only one of the two can gain access to the resource at a time. Terminal capabilities exchange-This is one of the most important functions of the H.245 protocol. The two most important capabilities are the supported audio codecs and the basic audio calls. Logical channel signaling-This indicates a one-way audio stream. With H.323 version 2, it is possible to open and close logical channels in the middle of a call. Because H.245 messages are independent of the H.225 signaling, a call can still be connected in H.225 even if no logical channels are open. This is typical with such features as hold, transfer, and conference. DTMF relay-Because voice networks typically do not carry DTMF tones inband because of compression issues, these tones are carried on the signaling channel. Ensure that the type of DTMF relay configured on your gateway is compatible with your gatekeeper.
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/voice_troubleshooting/old/vts_h323.html#wp106
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