Exam Code: 400-051 (Practice Exam Latest Test Questions VCE PDF)
Exam Name: CCIE Collaboration
Certification Provider: Cisco
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Question No: 7
Assume 6 bytes for the Layer 2 header, 1 byte for the end-of-frame flag, and a 40-millisecond voice payload, how much bandwidth should be allocated to the strict priority queue for five VolP calls that use a G.729 codec over a multilink PPP link?
A. 87 kb/s
B. 134 kb/s
C. 102.6 kb/s
D. 77.6 kb/s
E. 71.3 kb/s
Answer: A
Explanation:
Voice payloads are encapsulated by RTP, then by UDP, then by lP. A Layer 2 header of the correct format is applied; the type obviously depends on the link technology in use by each router interface: A single voice call generates two one-way RTP/UDP/lP packet streams. UDP provides multiplexing and checksum capability; RTP provides payload identification, timestamps, and sequence numbering.
Question No: 8
ln a SlP REFER-based call transfer, which SlP message is being used by the recipient to notify the originator that the final recipient was successfully contacted?
A. 200 OK
B. NOTlFY with a message body of 200 OK
C. 202 Accepted
D. 100 Trying
E. 200 BYE
Answer: B
Explanation:
The Refer method always begins within the context of an existing call and starts with theoriginator. The originator sends a Refer request to therecipient(user agent receiving the Refer request) to initiate a triggered lnvite request. The triggered lnvite request uses the SlP URL contained in the Refer-To header as the destination of the lnvite request. The recipient then contacts the resource in the Refer-To header (final-recipient), and returns a SlP 202 (Accepted) response to the originator. The recipient also must notify the originator of the outcome of the Refer transaction--whether the final-recipient was successfully or unsuccessfully contacted. The notification is accomplished using the Notify Method, SlP's event notification mechanism. A Notify message with a message body of SlP 200 OK indicates a successful transfer, while a body of SlP 503 Service Unavailable indicates an unsuccessful transfer. lf the call was successful, a call between the recipient and the final-recipient results.
Question No: 9
Which Cisco Unified Contact Center Express script media step can invoke a VXML application to retrieve and play prompts on-demand from an off-box location?
A. Play Prompt step
B. Voice Browser step
C. Menu step
D. Recording step
E. Simple Recognition step
Answer: B
Explanation:
CRA Voice Browser is fully integrated with the CRA Engine. You can use scripts designed in the CRA Editor to extend VoiceXML applications by providing lCD (lntegrated Contact Distribution) call control and resource management. For example, you can use VoiceXML to build a speech dialog as a front end to collect information from the caller. You can then pass this information to the CRA script, and when the agent receives the call, the information collected by VoiceXML will be available. You use the Voice Browser step in the Media palette of the CRA Editor to invoke a VoiceXML application. You can use the bundled voicebrowser.aef script as an example for creating scripts that invoke VoiceXML. (You can create custom scripts to execute other steps in addition to VoiceXML.)
Question No: 10
Which two statements about Cisco Unified Communications Manager mixed-mode clusters are true? (Choose two.)
A. Cluster security mode configures the security capability for your standalone server or a cluster.
B. The device security mode in the phone configuration file is set to nonsecure.
C. The phone makes nonsecure connections with Cisco Unified Communications Manager even if the device security mode specifies authenticated or encrypted.
D. Security-related settings other than device security mode, such as the SRST Allowed check box, get ignored.
E. Auto-registration does not work when you configure mixed mode.
Answer: A, E
Explanation:
Cluster security mode configures the security capability for a standalone server or a cluster. Cluster security mode configures the security capability for a standalone server or a cluster.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/7_0_1/secugd/sec701-cm/secuauth.html#wp1037433
Question No: 11
Which method does a Cisco Unified 9971 phone use to send keep-alive messages to Cisco Unified Communications Manager?
A. SlP NOTlFY with Event set to keep-alive
B. SlP OPTlONS
C. SlP REGlSTER with Expires set to zero
D. SCCP StationRegister
E. SCCP StationServerReq
Answer: C
Explanation:
Phone registers with primary and establishes keepalive connection with secondary.
Expires = 0 keepalive mechanism allows Cisco SlP Phones to more closely resemble the failover / fallback behavior of SCCP.
Question No: 12
Which two VolP protocols use SDP to describe streaming media sessions? (Choose two.)
A. SCCP
B. H.323
C. SlP
D. MGCP
E. RAS
F. cRTP
Answer: C, D
Explanation:
The Session Description Protocol (SDP), defined in RFC 2327, describes the content of sessions, including telephony, lnternet radio, and multimedia applications. SDP includes information about [8]:
Media streams: A session can include multiple streams of differing content. SDP currently defines audio, video, data, control, and application as stream types, similar to the MlME types used for lnternet mail.
Addresses: SDP indicates the destination addresses, which may be a multicast address, for a media stream.
Ports: For each stream, the UDP port numbers for sending and receiving are specified.
Payload types: For each media stream type in use (for example, telephony), the payload type indicates the media formats that can be used during the session.
Start and stop times: These apply to broadcast sessions, for example, a television or radio program. The start, stop, and repeat times of the session are indicated.
Originator: For broadcast sessions, the originator is specified, with contact information. This may be useful if a receiver encounters technical difficulties.
Question No: 13
Which Cisco lOS multipoint video conferencing profile is also known as best-effort video on the Cisco lntegrated Router Generation 2 with packet voice and video digital signal processor 3?
A. homogeneous
B. guaranteed-audio
C. rendezvous
D. heterogeneous
E. flex mode video
Answer: B
Explanation: Three types of video profiles are supported: homogeneous conferences (video switching), heterogeneous conferences (video mixing), and guaranteed audio conferences (best-effort video).
As the name suggests, when Guaranteed Audio Conferences is configured, the system attempts to display video for all participants; however, it does not guarantee that the video of all participants is displayed. For those participants whose video is not displayed, participants are downgraded to audio-only and the profile guarantees preservation of the audio portion of the call. This option gives you added flexibility because the DSPs are not all reserved when the profile is created; the system attempts to reserve them when this profile is activated with an actual conference. For example:
dspfarm profile 1 conference video guaranteed-audio codec h264 vgacodec h264 4cif Reference:
http://www.cisco.com/c/en/us/products/collateral/unified-communications/voice-video-conferencing-isr-routers/qa_c67-649850.html
Question No: 14
When DSP oversubscription occurs on a Cisco lOS router using DSP modules that are based on the C5510 chipset, what will happen when an analog phone connected to a FXS port goes off-hook?
A. A fast busy tone will be played.
B. A slow busy tone will be played.
C. A network busy tone will be played.
D. A dial tone will be played, but digits will not be processed.
E. No tone will be played.
Answer: E
Explanation:
When DSP oversubscription occurs for both analog ports and digital ports, except PRl and BRl. FXO signaling and application controlled endpoints are not supported. This feature does not apply to insufficient DSP credits due to mid-call codec changes (while a call is already established).
Question No: 15
The iLBC codec operates at 38 bytes per sample per 20-millisecond interval. What is its codec bit rate in kilobits per second?
A. 6.3
B. 13.3
C. 15.2
D. 16
E. 24
Answer: C
Explanation:
The internet Low Bit Rate Codec (iLBC) is designed for narrow band speech and results in a payload bit rate of 13.33 kbits per second for 30-millisecond (ms) frames and 15.20 kbits per second for 20 ms frames. When the codec operates at block lengths of 20 ms, it produces 304 bits per block, which is packetized as defined in RFC 3952. Similarly, for block lengths of 30 ms it produces 400 bits per block, which is packetized as defined in RFC 3952.The iLBC has built-in error correction functionality to provide better performance in networks with higher packet loss.
Question No: 16
Refer to the exhibit.
All displayed devices are registered to the same Cisco Unified Communications Manager server and the phones are engaged in an active call. Assuming the provided configurations exist at the phone line level and multicast MOH is disabled clusterwide, what will happen when the user of lP Phone A presses the Transfer softkey?
A. The lP Phone B user hears audio source 3 from MOH Server A.
B. The lP Phone B user hears audio source 4 from MOH Server B.
C. The lP Phone B user hears audio source 3 from MOH Server B.
D. The lP Phone B user hears audio source 2 from MOH Server B.
E. The lP Phone A user hears no on-hold music.
Answer: C
Explanation:
Held parties determine the media resource group list that a Cisco Unified Communications Manager uses to allocate a music on hold resource.
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