Cisco Certified Network Professional (CCNP) Collaboration — Practice Set 1

Question 1

A company wants to move its on-premises CUCM deployment to a cloud-only collaboration solution while keeping existing Cisco IP phones registered. Which deployment model allows existing SIP or SCCP phones to register to Cisco's cloud platform?

Accepted Answer

Cisco Webex Calling with Dedicated Instance (formerly known as Hosted Collaboration Solution / HCS) provides a private, dedicated tenant of Cisco Unified Communications Manager hosted in Cisco's data centers. This supports the full range of existing Cisco SIP and SCCP endpoints and features that on-premises CUCM supports. Standard multi-tenant Webex Calling uses a different phone registration model and does not support legacy SCCP devices. Cisco Meeting Server is a conferencing platform, not a phone registration system.

Question 2

A telephony engineer is configuring CUCM for globalized dial plan design. In a globalized dial plan, how should directory numbers (DNs) and translation patterns be formatted to support consistent international dialing?

Accepted Answer

In a CUCM globalized dial plan, all directory numbers are stored in E.164 format (e.g., +14085551234) with the leading + to ensure global uniqueness and consistent routing regardless of calling location. Translation patterns are used to normalize inbound calls from gateways (stripping the trunk access code, adding the + and country code) and to transform outbound calls (removing the + and prepending the appropriate PSTN access digit string for the egress gateway). This eliminates the need for location-specific dial plans.

Question 3

A branch office has a single ISDN PRI connection to the PSTN and a WAN link to headquarters where CUCM is deployed. The WAN link is unreliable. Which feature provides continued PSTN calling for the branch if the WAN link fails?

Accepted Answer

Cisco Unified SRST (Survivable Remote Site Telephony) is configured on the branch IOS XE gateway and provides fallback call control when the WAN link to CUCM is unavailable. During WAN failure, phones detect the loss of CUCM registration and automatically re-register to the SRST gateway. SRST provides basic call control (internal calls, PSTN calls via the local PRI) until the WAN link recovers and phones re-register to CUCM. CUBE is a session border element, not a fallback call controller. CER handles emergency call routing, not WAN failover.

Question 4

A voice engineer needs to choose a codec for a branch office connected by a 128 kbps WAN link that will carry up to 4 simultaneous voice calls. Which codec BEST balances voice quality and bandwidth efficiency for this scenario?

Accepted Answer

G.729 uses 8 kbps audio payload with a 20 ms packetization interval, resulting in approximately 24 kbps total bandwidth per call including RTP/UDP/IP headers. Four simultaneous G.729 calls consume approximately 96 kbps, which fits within the 128 kbps WAN link and leaves headroom for signaling traffic. G.711 (both µ-law and a-law) uses 64 kbps audio payload (~80 kbps total), making 4 calls require 320 kbps — well beyond the available bandwidth. G.722 also uses 64 kbps and is a wideband codec not suitable for bandwidth-constrained WAN links.

Question 5

A collaboration architect is designing a CUCM cluster with 5000 IP phones across three sites. Which CUCM cluster roles and design elements ensure high availability for call processing?

Accepted Answer

A CUCM cluster has exactly one Publisher node that acts as the primary database master for configuration replication to all Subscriber nodes. Phones should be registered to Subscriber nodes (call-processing nodes), not the Publisher, so that Publisher maintenance or failure does not interrupt active calls. CUCM supports primary, secondary (and tertiary in some versions) Subscriber failover per phone. For remote sites without dedicated Subscribers, SRST on the local gateway provides fallback. There is no active-active multi-Publisher CUCM design; the Publisher is a single instance.

Question 6

A network engineer is examining a SIP call setup trace. Which sequence of SIP messages correctly represents a successful call setup followed by call teardown?

Accepted Answer

The standard SIP call flow is: INVITE (caller initiates the session) → 180 Ringing (callee's UA alerts the user, provisional response) → 200 OK (callee answers) → ACK (caller acknowledges the 200 OK, completing the three-way handshake) → [media via RTP] → BYE (either party ends the call) → 200 OK (confirms the BYE). Option A shows 100 Trying (server acknowledgment of the INVITE) but skips 180 Ringing. Option C includes REGISTER which is a separate registration flow, not part of a call setup sequence. Option D has 200 OK before 180 Ringing, which is incorrect — provisional responses precede the final 200 OK.

Question 7

An engineer is troubleshooting a one-way audio issue on a SIP call through a CUBE. The SDP in the 200 OK response shows the correct IP and RTP port for the far end. However, audio flows only from callee to caller. What is the MOST likely cause?

Accepted Answer

One-way audio where media flows in only one direction (callee to caller but not caller to callee) is a classic symptom of an asymmetric firewall or ACL blocking RTP UDP traffic in one direction. The RTP port range used by Cisco endpoints and gateways is 16384–32767. If the firewall blocks UDP traffic from the caller's network toward the callee's RTP port, the callee receives no audio from the caller but the caller can still hear the callee. An SDP 'a=recvonly' attribute would also cause this symptom and is a valid alternative cause — but firewall blocking is the most common real-world cause for this specific symptom pattern. RTCP is for statistics reporting and its absence does not affect audio flow.

Question 8

A Cisco IP phone needs to download its configuration file after being connected to the network. In which order does the phone contact services to obtain its configuration?

Accepted Answer

Cisco IP phone initialization follows this sequence: (1) the phone obtains an IP address via DHCP, which also returns DHCP option 150 (TFTP server address) or option 66; (2) the phone contacts the TFTP server and downloads its configuration file (SEP<MAC>.cnf.xml for SIP phones, or the equivalent for SCCP); (3) the configuration file contains the CUCM node addresses; (4) the phone registers to CUCM using SIP (REGISTER) or SCCP. CDP is used for voice VLAN assignment and PoE negotiation but does not carry CUCM addresses. Direct SIP REGISTER without a TFTP config stage is not the standard Cisco phone boot sequence.

Question 9

An engineer is reviewing SRTP configuration on CUCM. Which key exchange mechanism does CUCM use by default for SRTP media encryption between Cisco IP phones?

Accepted Answer

Cisco CUCM uses SDES (SDP Security Descriptions, RFC 4568) for SRTP key exchange. The SRTP master key and salt are embedded in the SDP 'a=crypto:' attribute within the SIP INVITE and 200 OK messages. To protect the keys in transit, the SIP signaling must be encrypted with SIP TLS (port 5061). DTLS-SRTP (RFC 5764) is used in WebRTC environments. MIKEY and IKEv2 are not used by Cisco CUCM for standard SRTP. CUCM must be in Mixed Mode (CTL installed) to enable SRTP with SDES.

Question 10

An engineer captures a SIP trace and observes that the SDP offer in the INVITE contains 'a=sendrecv' and two 'm=' lines — one for audio (m=audio 16388 RTP/SAVP 18 0) and one for video (m=video 16390 RTP/SAVP 97). The 200 OK responds with 'm=audio 49152 RTP/SAVP 0' and 'm=video 0 RTP/SAVP 97'. What has the callee negotiated?

Accepted Answer

In SDP negotiation, a port value of 0 in the m= line indicates that the media stream is rejected (inactive). The callee's 200 OK with 'm=audio 49152 RTP/SAVP 0' accepts audio using RTP payload type 0 (G.711 µ-law). The 'm=video 0 RTP/SAVP 97' with port 0 means the callee is declining the video stream. The INVITE's audio offer listed payload 18 (G.729) first and 0 (G.711) second — the callee selected payload type 0 (G.711), the second offered codec. This is standard SDP offer/answer negotiation per RFC 3264.

Question 11

A voice engineer is configuring a Cisco CUBE to connect a SIP trunk from a service provider to CUCM. Which IOS XE command defines the VoIP dial-peer that matches outbound calls toward the service provider SIP trunk?

Accepted Answer

A VoIP dial-peer ('dial-peer voice 200 voip') is used for SIP trunk connectivity on CUBE. The 'session protocol sipv2' command specifies SIP as the signaling protocol. 'session target sip-server' points to the configured SIP server (the service provider or CUCM) defined under 'voice service voip / sip'. The 'destination-pattern .T' matches any dial string. The 'codec' statement specifies the preferred codec. A POTS dial-peer (option A) is for physical PSTN interfaces (PRI/FXO), not SIP trunks. Option C configures the global SIP binding, not a dial-peer. Option D is a gateway timer, not dial-peer configuration.

Question 12

A collaboration engineer is configuring a hardware conference bridge on a Cisco IOS XE DSP-enabled gateway. Which command block registers a dspfarm conference profile with CUCM?

Accepted Answer

Cisco IOS XE DSPfarm conference bridges are configured using the 'dspfarm profile' command. The profile must specify the codec, maximum sessions, and 'associate application SCCP' to register the resource with CUCM via SCCP (Skinny). The SCCP configuration ('sccp local', 'sccp ccm', 'sccp') establishes the SCCP connection between the gateway's DSP farm and CUCM, allowing CUCM to invoke the conference bridge for ad-hoc or meet-me conferences. Option B is an incorrect dial-peer approach. Options C and D are not valid IOS XE commands for conference bridge configuration.

Question 13

A CUCM call between a G.711 endpoint and a G.729 endpoint fails with the error 'No Common Codec'. A Media Termination Point (MTP) is already allocated. Which resource resolves the codec mismatch and allows the call to proceed?

Accepted Answer

When two endpoints negotiate incompatible codecs, a transcoder is required to convert the audio between codec formats in real time. A DSPfarm transcoder profile (configured as 'dspfarm profile N transcode') registered to CUCM via SCCP is inserted into the media path by CUCM when it detects a codec mismatch. An MTP (Media Termination Point) does NOT transcode — it only anchors media for features like hold/transfer with endpoints that do not support RFC 2833 DTMF relay or third-party call control; it does not perform codec conversion. Changing the region codec to G.711 forces G.711 everywhere but does not solve the scenario where the remote endpoint only supports G.729.

Question 14

An engineer is configuring Cisco CUBE (IOS XE 17.x) for a Webex Calling Local Gateway deployment. Which two commands are REQUIRED on the CUBE's VoIP dial-peer toward Webex to ensure proper Webex Calling LGW behavior? (Choose 2)

Accepted Answer

For Cisco Webex Calling Local Gateway on IOS XE CUBE, two specific commands are required on the outbound VoIP dial-peer toward Webex: (1) 'voice class stun-usage <tag>' — enables ICE/STUN media negotiation required by Webex Calling for NAT traversal and media path establishment through Webex's TURN servers; (2) 'no supplementary-service sip refer' — disables CUBE's default behavior of passing SIP REFER messages (for call transfer) natively, instead requiring CUBE to consume REFER and issue new INVITE dialogs, which Webex Calling requires. These are documented in the official Cisco Webex Calling LGW configuration guide. 'codec transparent' and 'session protocol sipv2' are used in general CUBE configuration but are not the Webex-specific requirements. 'dtmf-relay rtp-nte' is a standard DTMF configuration, not Webex-specific.

Question 15

A CUCM administrator needs to allow a user to log into any Cisco IP phone on campus and have their personal line, speed dials, and services follow them. Which CUCM feature provides this capability?

Accepted Answer

Cisco CUCM Extension Mobility allows a user to log in to any Extension Mobility-enabled Cisco IP phone using a user ID and PIN. Upon login, the phone downloads the user's Device Profile (their directory number, line appearances, speed dials, and subscribed phone services). The host phone reverts to its default profile on logout. This is designed for hot-desking and shared workspace environments. Call Park is for holding calls at a monitored extension. Single Number Reach extends calls to a mobile. Cisco Jabber is a separate soft-client application.

Question 16

A CUCM administrator needs to add 500 new employees to CUCM with phones, user accounts, and voicemail pilot numbers. Which CUCM tool is designed for this type of bulk provisioning task?

Accepted Answer

The Cisco Bulk Administration Tool (BAT) is a CUCM Administration web interface feature that allows administrators to add, update, or delete large numbers of phones, users, user-device associations, lines, and other CUCM objects by uploading a CSV file formatted according to BAT's template specifications. BAT significantly reduces the time required for bulk provisioning compared to adding devices one at a time. RTMT is a monitoring tool. The CUCM CLI provides system-level administration (show/set commands) but not phone/user provisioning. Prime Collaboration Provisioning is an external tool with broader scope.

Question 17

A CUCM administrator observes that an outbound call from extension 5001 to PSTN number 9-408-555-1234 is being routed to the wrong gateway. The administrator checks the route pattern 9.1[2-9]XX[2-9]XXXXXX and the route list. Which CUCM call routing component allows the administrator to define multiple gateways and a selection algorithm (top-down, circular) for PSTN egress?

Accepted Answer

In CUCM call routing, a Route Group contains one or more PSTN gateways or SIP trunks and defines how they are selected: top-down (always tries the first gateway first) or circular (round-robin). Route Groups are assigned to a Route List, which in turn is assigned to a Route Pattern. This hierarchy (Route Pattern → Route List → Route Group → Gateway/Trunk) allows flexible load balancing and failover across multiple PSTN egress points. A Route Pattern by itself only points to a single destination (a route list or gateway). Translation patterns modify digits. Calling Search Spaces control which partitions a device can reach.

Question 18

A CUCM administrator configures Single Number Reach for a mobile worker. The worker's desk phone (DN 5100) should simultaneously ring their mobile number (408-555-9876) when calls arrive. Which CUCM feature configuration achieves this?

Accepted Answer

CUCM Single Number Reach (SNR) uses a Remote Destination Profile (RDP), which is a virtual device associated with a user's desk phone DN. A Remote Destination (the mobile number, formatted for outbound PSTN dialing) is added to the RDP with SNR enabled. When a call arrives at the desk DN, CUCM simultaneously places an outbound call to the configured remote destination. The user can answer on either device, and CUCM allows mid-call mobility (moving the active call between desk phone and mobile). Hunt groups distribute calls across agents, not the same user on multiple devices.

Question 19

A CUCM 14.x administrator needs to ensure that calls from a partner company (SIP trunk, partition PARTNER-PT) can reach internal extensions (partition INTERNAL-PT) but cannot call external PSTN numbers (partition PSTN-PT). Which configuration CORRECTLY enforces this restriction?

Accepted Answer

In CUCM, a Calling Search Space (CSS) is an ordered list of partitions that defines which partitions a calling device or line can reach. To restrict the SIP trunk (representing the partner company) from calling PSTN numbers: create a CSS that contains only INTERNAL-PT (the partition holding internal DNs) but not PSTN-PT (the partition containing PSTN route patterns). Assign this CSS to the SIP trunk. The trunk's calls will be allowed to match route patterns/DNs in INTERNAL-PT but will fail to match anything in PSTN-PT, resulting in call rejection for outbound PSTN attempts. Option B incorrectly adds PSTN-PT to the CSS. A translation pattern block is not the appropriate mechanism — CUCM uses CSS/partition for class of service.

Question 20

A company wants to use Cisco Webex Calling but needs to keep its existing on-premises PBX for some legacy analog devices. They want Webex Calling users to be able to call legacy PBX extensions and vice versa. Which Webex Calling connectivity option supports this by using an on-premises IOS XE CUBE as the interconnect?

Accepted Answer

Webex Calling Local Gateway (Premises-Based PSTN / LGW) uses a Cisco IOS XE CUBE deployed on-premises and registered to the Webex Calling cloud as a trunk. This CUBE can interconnect the on-premises PBX (via SIP or ISDN PRI), existing analog devices, and local PSTN circuits with Webex Calling users in the cloud. Cisco Calling Plans and CCP are cloud-based PSTN connectivity options that do not involve an on-premises gateway. Webex Edge for Devices is for room systems, not PSTN/PBX connectivity.

Question 21

A Webex Calling administrator uses Cisco Control Hub to manage locations and calling settings. A new office location in Dallas needs to be provisioned. Which Control Hub configuration steps are required to enable users at this location to make outbound PSTN calls via a Cisco Calling Plan?

Accepted Answer

In Webex Control Hub, Locations represent physical offices or sites and are the fundamental unit for PSTN configuration. The administrator must: (1) create a Location for Dallas; (2) assign a PSTN connection type to the location — in this case, a Cisco Calling Plan (Cisco-managed SIP trunks); (3) add telephone numbers (DIDs) from the Cisco Calling Plan inventory to the Dallas location; (4) assign those numbers to Webex Calling users. There is no CUCM involved in a pure Webex Calling deployment, and PSTN does not flow automatically — explicit location and PSTN assignment are required.

Question 22

A company using Webex Calling LGW (IOS XE CUBE) needs to ensure that call transfers initiated by Webex Calling endpoints are handled correctly through the LGW toward the PSTN. Which behavior does 'no supplementary-service sip refer' enforce on the LGW dial-peer toward Webex?

Accepted Answer

By default, IOS XE CUBE passes SIP REFER messages through to the far-end (transparent REFER), which can cause problems when the PSTN gateway or upstream provider does not support SIP REFER-based call transfer. The command 'no supplementary-service sip refer' on the Webex-facing dial-peer disables this pass-through behavior. Instead, the CUBE consumes the REFER, terminates the existing call leg, and issues a new SIP INVITE to the transfer target. Webex Calling's LGW configuration guides require this command specifically because Webex generates REFER for transfers, and the CUBE must handle the transfer — not pass it to the PSTN provider, which may not support it.

Question 23

A CUCM-to-Webex Calling migration is underway. Some users are on Webex Calling; others remain on CUCM. An engineer is configuring interoperability so CUCM users can dial Webex Calling users via a 4-digit extension (8XXX range) routed through a CUBE LGW. Which design ensures CUCM routes 8XXX calls to the Webex Calling tenant?

Accepted Answer

For CUCM-to-Webex Calling interoperability, the recommended design is: (1) CUCM route pattern 8XXX (in the appropriate partition) points to a SIP trunk connecting CUCM to the CUBE LGW; (2) the CUBE LGW has dial-peers matching 8XXX inbound from CUCM, normalizes the dialed number to the corresponding Webex Calling E.164 number via a voice translation profile, and forwards the call via the registered Webex Calling LGW trunk to the Webex cloud; (3) Webex Calling matches the +E.164 to the user's Webex Calling number and rings the Webex endpoint. This avoids routing over the PSTN (which would incur charges and add latency) and provides a direct IP path between the two calling systems.

Question 24

A network engineer is configuring QoS for voice traffic on a WAN router. Which DSCP marking should be applied to voice RTP media packets?

Accepted Answer

DSCP EF (Expedited Forwarding, decimal 46, binary 101110) is the RFC 3246 per-hop behavior assigned to voice RTP media. EF traffic is placed in the Low Latency Queue (LLQ) on Cisco routers, which guarantees strict-priority treatment: the queue is always serviced before other queues, providing the lowest latency and jitter. DSCP CS3 (24) is the recommended marking for call signaling (SIP, SCCP) per Cisco QoS Baseline. AF41 is recommended for interactive video conferencing. BE (0) is default best-effort with no QoS guarantees.

Question 25

A security administrator is enabling encrypted signaling and media for CUCM. Which combination of technologies provides both SIP signaling encryption AND media encryption for calls between Cisco IP phones and CUCM?

Accepted Answer

To fully secure CUCM calls: (1) SIP TLS on port 5061 encrypts the SIP signaling messages between the IP phone and CUCM (replacing SIP over TCP/UDP on port 5060); (2) SRTP encrypts the RTP media stream between endpoints using keys exchanged via SDES in the SDP body of the TLS-protected SIP messages. Cisco CUCM uses SDES (not DTLS) for SRTP key exchange. Enabling CUCM Mixed Mode (by installing the Certificate Trust List, CTL) is a prerequisite — it enables the security capabilities on the cluster. HTTPS protects only the admin web interface. IPsec is not a standard CUCM endpoint security mechanism. DTLS is used in WebRTC, not native Cisco CUCM.

350-801 CLCOR — Cisco Certified Network Professional (CCNP) Collaboration