Cisco 350-801 CLCOR cheat sheet

A single Cisco Unified Communications Manager (CUCM) cluster supports a maximum of 20,000 users; larger designs use multiple clusters connected with Intercluster Lookup Service (ILS).

CUCM OVA templates are sized by user count (1,000 / 2,500 / 7,500 / 10,000 / 15,000 / 20,000) and fix vCPU, RAM, and disk; the deployed OVA must match the supported UC version and ESXi matrix.

DHCP Option 150 returns a list of TFTP server IP addresses to IP phones; Option 66 returns a single TFTP server name and is used as an alternative.

Mobile and Remote Access (MRA) places Expressway-C inside the enterprise and Expressway-E in the DMZ; the two are joined by a traversal zone that carries SIP and media through the firewall.

Smart Licensing runs in evaluation mode for 90 days; after registration expires there is an out-of-compliance grace period before enforcement blocks new registrations.

Every CUCM node must synchronize to a reliable NTP source; time skew invalidates certificates, breaks database replication, and corrupts CDR timestamps.

The first node installed is the publisher and is the master database; subscribers replicate from it, and configuration changes are blocked while the publisher is down even though established calls continue.

Quality of Service trust is applied at the access-switch port toward the phone; voice bearer is marked DSCP EF (46) and call signaling CS3 (24).

SIP requests used in CUCM call flows include INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS, PRACK, UPDATE, INFO, REFER, and SUBSCRIBE/NOTIFY.

SIP response classes: 1xx provisional (100 Trying, 180 Ringing, 183 Session Progress), 2xx success (200 OK), 3xx redirect, 4xx client error (401/407 authentication, 486 Busy Here, 488 Not Acceptable Here), 5xx server error (503), 6xx global failure (603 Decline).

SDP offer/answer uses m= for media type, port, and RTP profile, a=rtpmap to map payload type to codec and clock rate, and c= for the connection address.

RTP media travels on even UDP ports in the range 16384-32767; RTCP uses the next higher odd port and reports jitter, packet loss, and round-trip delay.

G.711 (PCMU/PCMA) carries a 64 kbps payload and consumes about 80 kbps on the wire with IP/UDP/RTP headers at 20 ms packetization; G.729 carries an 8 kbps payload and about 24-31 kbps on the wire.

G.722 is a 64 kbps wideband codec sampled at 16 kHz that provides 7 kHz of audio bandwidth; Opus is variable rate and used for WebRTC-based endpoints.

Signaling ports: SCCP phones to CUCM over TCP 2000 (TLS 2443 for secure SCCP), SIP over UDP/TCP 5060 and TLS 5061, H.323 call signaling TCP 1720 with RAS UDP 1719, MGCP gateway control UDP 2427 with call agent UDP 2727.

DTMF transport options are RFC 4733 (formerly RFC 2833) telephone-event RTP packets, inband audio tones, SIP INFO messages, and KPML for SCCP endpoints; RFC 4733 is preferred across compressed codecs.

Phone boot sequence: DHCP (Option 150) gives the TFTP address, the phone downloads its configuration file from TFTP, then registers to a CUCM node over SCCP or SIP.

The phone configuration file is named SEP followed by the MAC address with no separators and a .cnf.xml suffix, for example SEP001122334455.cnf.xml.

A device pool defines CUCM group, region, location, date/time group, SRST reference, network locale, calling search space, and media resource group list for a set of devices.

Auto-registration assigns unprovisioned phones a directory number from the configured range plus the auto-registration device pool and calling search space; it is disabled by default in secure clusters.

Bulk Administration Tool (BAT) imports and exports phones, users, lines, and templates from CSV files; jobs are validated before insert and can be rolled back.

Extension Mobility downloads the user's device profile to a shared phone after login and restores the default profile on logout; Cisco Extension Mobility uses a dedicated service on CUCM.

CAPF (Certificate Authority Proxy Function) issues and manages Locally Significant Certificates on phones so that TLS signaling and SRTP media encryption can be enabled.

A phone button template defines line, speed-dial, and feature buttons; a softkey template defines the softkeys available during call states.

A partition groups directory numbers and patterns; a calling search space (CSS) is an ordered list of partitions, and CUCM searches the CSS top-down and uses the first matching pattern.

Dial-plan hierarchy: route pattern points to a route list, the route list contains ordered route groups, and each route group contains gateways or trunks with a distribution algorithm.

Route group distribution algorithms are top-down, circular, and longest idle; if the selected device fails, the route list moves to the next route group.

Wildcards in CUCM patterns: @ matches a complete directory number, X matches any single digit 0-9, ! matches one or more digits, [0-9] defines a digit set, and a hyphen defines a numeric range.

A translation pattern matches dialed digits, rewrites called or calling numbers, and then routes the transformed number using the calling search space assigned to the translation pattern.

Hunt pilot points to a hunt list, which contains line groups; line-group members receive calls according to the configured algorithm (top-down, circular, longest idle, broadcast).

Location-based call admission control defines the audio and video bandwidth available between sites; CUCM rejects a call when the requested bandwidth exceeds the remaining location budget.

Forced Authorization Codes validate a caller's privilege level before the call completes; Client Matter Codes tag the call record for billing or client tracking.

Media resource selection follows the device's media resource group list (MRGL): MRGs are searched in order, and within each MRG the resources are tried in the configured order.

A Media Termination Point (MTP) relays RTP and is inserted for DTMF interworking, SIP early offer, protocol interworking, and supplementary services between differing signaling types.

A transcoder converts between codecs (for example G.729 to G.711) using DSP resources and registers with CUCM over SCCP; it is inserted when endpoints share no common codec.

Conference bridges are either hardware (PVDM DSPs on a voice gateway) or software (Cisco Conference Bridge on a CUCM node); ad hoc conferences use the phone's Conference softkey, meet-me conferences use a dialable number.

Music on hold can be unicast (one stream per held call) or multicast (one stream to a multicast address and port shared by many listeners, which saves WAN bandwidth).

The Annunciator is a software media resource that plays announcements and tones, such as the welcome greeting or a message played to callers.

A region defines the maximum codec used between devices assigned to different regions (for example G.722 within a region, G.729 between regions), while a location limits the bandwidth those calls may consume.

An RSVP agent is a media resource that reserves bandwidth along the call path to provide call admission control for a location pair.