Free CCIE Collaboration Practice Questions

Question 1

Reception phones must retain domestic PSTN dialing through access digit 9 but must not place international calls. Their effective calling search space contains PT-Access and PT-Domestic, not PT-International. Direct dialing of an international +E.164 number is blocked, yet dialing that destination through 9 succeeds. Digit analysis shows:

Translation pattern: 9.! in PT-Access
Digit treatment: discard PreDot, then prefix +
Translation pattern CSS: CSS-Unrestricted
Use Originator's Calling Search Space: unchecked
CSS-Unrestricted: PT-Domestic, PT-International
CSS-Domestic: PT-Domestic

All permitted normalized domestic destinations are in PT-Domestic. How should the international-access gap be closed while preserving 9-prefixed domestic dialing?

Accepted Answer

Assign CSS-Domestic to the 9.! translation pattern.

Answer

Remove the 9.! pattern from PT-Access.

Answer

Move 9.! to PT-Domestic without changing its translation CSS.

Answer

Set the egress called-party number type to National.

Question 2

During an IVR acceptance test, speech works in both directions through CUBE, but the IVR ignores keypad input. The completed SDP exchanges negotiate PCMU for speech and these telephone-event mappings:

CUCM-facing leg: a=rtpmap:101 telephone-event/8000
Carrier-facing leg: a=rtpmap:110 telephone-event/8000
Both legs: telephone-event range 0-15

A capture at the carrier shows the packets for a pressed # key arriving with RTP payload type 101 and event value 11. The carrier requires RTP telephone events using its negotiated payload type; its mapping cannot be changed. What should the interworking repair accomplish?

Accepted Answer

Use payload type 110 on the carrier leg and retain event value 11.

Answer

Use payload type 101 on both call legs without changing the event value.

Answer

Replace telephone-event packets with in-band G.711 tones on the carrier leg.

Answer

Use payload type 110 and change the # event from 11 to 10.

Question 3

'Every sentence sounds clear, but we keep talking over each other.' That is the complaint on a G.711 intersite call. Measurements collected during the complaint show MOS-LQK averaging 4.3 with a minimum of 4.1, packet loss of 0.1%, jitter of 6 ms, no late-packet discards, and measured mouth-to-ear one-way delay of 270 ms. The enterprise's one-way voice-delay target is 150 ms. Which assessment best reconciles the complaint with these measurements?

Accepted Answer

Conversational delay is excessive despite good estimated listening quality.

Answer

Late-packet loss is the dominant impairment, concealed by the codec.

Answer

An undersized jitter buffer is the main cause of the interruptions.

Answer

The call meets its voice target because the MOS-LQK exceeds 4.0.

Question 4

An MRA call connects: the remote user hears the office employee, but the employee cannot hear the remote user. Media is anchored on Expressway-E; ICE optimization is not in use. Expressway-E has private address 10.20.30.40 and a static one-to-one public mapping of 203.0.113.40. A capture confirms that Expressway-E generates SDP advertising 10.20.30.40 as the audio receive address, and that SDP reaches the client unchanged. The client sends media toward this unroutable private address. The negotiated public UDP port is already permitted and correctly translated by the firewall. Where should the engineer make the correction?

Accepted Answer

Configure Expressway-E's static NAT address as 203.0.113.40.

Answer

Permit all UDP media ports on the perimeter firewall.

Answer

Replace the Expressway-E certificate with one containing the public IP.

Answer

Change the codec preference order for calls traversing Expressway.

Question 5

A recording rollout uses the Built-in Bridge on supported Cisco phones, not gateway-based media forking. Normal calls have two-way audio. CUCM successfully establishes each recording session with the recording server, and phone diagnostics confirm that the recording streams are being sent, but the server receives no RTP. The firewall currently permits the negotiated recording-media ports only when the source address belongs to the CUCM server subnet. Which additional source-to-destination flow is required for this recording architecture?

Accepted Answer

RTP from the phone voice VLANs to the recording server.

Answer

RTP from the CUBE media interfaces to the recording server.

Answer

RTP from the recording server to the CUCM subscribers.

Answer

RTP from the CUCM publisher to the recording server.

Question 6

A branch voice-admission policy caps aggregate IP-layer media traffic at 720 kb/s in each direction. Six active calls use IPv4; four additional calls would use IPv6. Every call carries G.711 at 64 kb/s with 20-ms packetization and one RTP stream per direction. Existing calls will retain IPv4. The accounting includes only the base IP, UDP, and RTP headers: no header compression, extensions, SRTP overhead, RTCP, signaling, or Layer 2 overhead. An alternate path can carry new calls that do not fit. Select the decision that admits the greatest number of new WAN calls without exceeding either directional limit.

Accepted Answer

Admit two new calls on the WAN and route the other two over the alternate path.

Answer

Admit three new calls on the WAN and route the remaining call over the alternate path.

Answer

Use the alternate path for all four new calls; the WAN budget is exhausted.

Answer

Keep all four new calls on the WAN; their aggregate codec rate is within the budget.

Question 7

A SIP call to 14085550123 has already matched inbound dial peer 100 on CUBE. All three outbound SIP dial peers below are operational, have reachable session targets, and can carry the call. DPG 70 is active. Relevant routing configuration is:

voice class dpg 70
 dial-peer 701 preference 1
 dial-peer 702 preference 2
!
dial-peer voice 100 voip
 destination dpg 70
!
dial-peer voice 701 voip
 destination-pattern 1415.......
!
dial-peer voice 702 voip
 destination-pattern 1408.......
!
dial-peer voice 703 voip
 destination-pattern 14085550123

No translation or other routing policy changes the called number or candidate selection. Which outbound selection occurs first?

Accepted Answer

701; its preference takes precedence over destination-pattern matching.

Answer

703; an exact destination-pattern match outranks the DPG.

Answer

702; both DPG membership and a matching destination-pattern are required.

Answer

701 or 702; all active DPG members receive calls equally.

Question 8

A board meeting has two mandatory controls: the service provider must not be able to decrypt its media, and an approved audiovisual recording must be retained on a company-managed endpoint. A cloud-generated transcript would be convenient but is optional. All attendees will use supported Webex desktop clients, and the organization has enabled Zero-Trust end-to-end encrypted meeting types. Which proposed configuration satisfies both mandatory controls?

Accepted Answer

Use Zero-Trust end-to-end encryption with approved local recording and no cloud transcription.

Answer

Use a transport-encrypted meeting with cloud recording and transcription.

Answer

Use Zero-Trust encryption but replace the required recording with an attendance report.

Answer

Use Video Mesh with cloud recording and transcription to keep the media on premises.

Question 9

A security review finds that a group of Cisco SIP phones has valid LSCs, accepted CTL/ITL trust lists, and successful TLS registration to CUCM. Their phone security profile is set to Authenticated. On a test call to an encryption-capable peer, the phones negotiate RTP/AVP rather than SRTP. The remaining call path already supports encrypted signaling and media. To meet the requirement for encrypted phone media, the engineer should:

Accepted Answer

Assign an encrypted phone security profile.

Answer

Set the SIP trunk to authenticated mode to match the phones.

Answer

Regenerate the CTL and ITL, retaining authenticated mode.

Answer

Renew the phones' LSCs, retaining the existing profiles.

Question 10

At peak attendance, a healthy Video Mesh cluster reaches capacity. Additional participants connect to cloud media resources; existing local participants remain connected with good quality. Control Hub identifies capacity overflow, and policy permits cloud media processing. How should operations handle this event?

Accepted Answer

Continue service and plan capacity for the observed overflow.

Answer

Pause new meetings until local capacity becomes available.

Answer

Restart the Video Mesh cluster to return cloud participants to local nodes.

Answer

Block the cloud media path to keep meetings on premises.

Free CCIE Collaboration Practice Questions

Domain 1: Infrastructure and Design 15% of exam

Question 1

Domain 2: Protocols and Endpoints 10% of exam

Question 2

Question 3

Domain 3: On-Premises Call Control 30% of exam

Question 4

Question 5

Domain 4: Voice Gateways and Session Border Controllers 10% of exam

Question 6

Domain 5: Cloud and Hybrid Services 25% of exam

Question 7

Domain 6: Media and QoS 10% of exam

Question 8

Question 9

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