Tips to Pass 400-051 Exam (14 to 23)

Answer: B

Explanation:

Admission Request (ARQ) and Location Request (LRQ) are the two H.225 Registration, Admission, Status (RAS) messages that trigger a gatekeeper to initiate the call routing decision process.

ARQ-Local zone messages that are sent by H.323 endpoints (usually gateways) to the Cisco gatekeeper. Gatekeepers receive ARQs from an endpoint if.

A local zone endpoint initiates a call. OR

A local zone endpoint request permission to admit an incoming call.

Answer: B

Explanation:

Under the policy map VOlP the CS3 value falls under the signal class-map, which has been allocated 5 percent of the bandwidth.

Answer: A

Explanation:

You can check the validity of your dial peer configuration by performing the following tasks:

u2022 lf you have relatively few dial peers configured, you can use theshow dial-peer voicecommand to verify that the configuration is correct. To display a specific dial peer or to display all configured dial peers, use this command. The following is sample output from theshow dial-peer voicecommand for a specific VolP dial peer:

router# show dial-peer voice 10

VoiceOverlpPeer10

tag = 10, dest-pat = \Q',

incall-number = \Q+14087',

group = 0, Admin state is up, Operation state is down

Permission is Answer,

type = voip, session-target = \Q',

sess-proto = cisco, req-qos = bestEffort,

acc-qos = bestEffort,

fax-rate = voice, codec = g729r8,

Expect factor = 10,lcpif = 30, VAD = disabled, Poor QOV Trap = disabled,

Connect Time = 0, Charged Units = 0

Successful Calls = 0, Failed Calls = 0

Accepted Calls = 0, Refused Calls = 0

Last Disconnect Cause is ""

Last Disconnect Text is ""

Last Setup Time = 0

u2022 To show the dial peer that matches a particular number (destination pattern), use theshow dialplan numbercommand. The following example displays the VolP dial peer associated with the destination pattern 51234:

router# show dialplan number 51234

Macro Exp.: 14085551234

VoiceOverlpPeer1004

tag = 1004, destination-pattern = \Q+1408555....',

answer-address = \Q',

group = 1004, Admin state is up, Operation state is up

type = voip, session-target = \Qipv4:1.13.24.0',

ip precedence: 0 UDP checksum = disabled

session-protocol = cisco, req-qos = best-effort,

acc-qos = best-effort,

fax-rate = voice, codec = g729r8, Expect factor = 10, lcpif = 30,

VAD = enabled, Poor QOV Trap = disabled Connect Time = 0, Charged Units = 0 Successful Calls = 0, Failed Calls = 0 Accepted Calls = 0, Refused Calls = 0

Last Disconnect Cause is "" Last Disconnect Text is "" Last Setup Time = 0

Matched: +14085551234 Digits: 7 Target: ipv4:172.13.24.0

Answer: A

Explanation:

Cisco Tomcat service, as the name suggests, is used by the Web Server of CUCM and helps display the administration, operating system, disaster recovery, and other GUl interfaces of CUCM. The service leverages a built-in CA for Tomcat in that it redirects the incoming HTTP requests to HTTPS using the default self-signed certificate.

Answer: C

Explanation:

All messages to and from a skinny device are preceded by either the words StationD or Stationlnit. StationD messages are sent from call manager to lP phone. Skinny message transmission such as this between the lP phone and CallManger occurs for every action undertaken by the lP phone, includinginitialization, registration, on-hook, off-hook, dialing of the digits, key press on the phone, and so much more.

Answer: A

Explanation:

Reference: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml

Answer: B

Explanation:

A typical H.245 exchange looks similar to below figure

After this exchange of messages, the two endpoints (EP) in this figure would be transmitting audio in each direction.

The number of message exchanges is numerous, each has an important purpose, but nonetheless takes time.

For this reason, H.323 version 2 (published in 1998) introduced a concept called Fast Connect, which enables a device to establish bi-directional media flows as part of the H.225.0 call establishment procedures. With Fast Connect, it is possible to establish a call with bi-directional media flowing with no more than two messages, like in figure 3.

Fast Connect is widely supported in the industry. Even so, most devices still implement the complete H.245 exchange as shown above and perform that message exchange in parallel to other activities, so there is no noticeable delay to the calling or called party.

Answer: C

Explanation:

Enterprise Medianet QoS Recommendations

Reference:

http://www.cisco.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND_40/QoSlntro_40.html#pgfld-61116

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.

Answer: B, C

Explanation:

iSAC-lnternet Speech Audio Codec (iSAC) is an adaptive wideband audio codec, specially designed to deliver wideband sound quality with low delay in both low and medium-bit rate applications. Using an adaptive bit rate of between 10 and 32 kb/s, iSAC provides audio quality approaching that of G.722 while using less than half the bandwidth. ln deployments with significant packet loss, delay, or jitter, such as over a WAN, iSAC audio quality is superior to that of G.722 due to its robustness. iSAC is supported for SlP and SCCP devices. The Cisco Unified Communications Manager lP Voice Media Streaming App (lPVMSApp), which includes Media Termination Point, Conference Bridge, Music on Hold Server, and Annunciator does not support iSAC. MGCP devices are not supported.