Cisco 300-830 CLCCE key terms

A Cisco-hosted, multi-tenant delivery of Unified Communications Manager where call processing, dial plans, and device registration run in Cisco's cloud instead of on customer premises hardware.

Cisco's cloud calling service built on a multi-tenant platform; it delivers PSTN access, call routing, and user features from the cloud and integrates with Webex App and registered endpoints.

A premises-based Cisco gateway registered to a cloud calling service that connects cloud-registered endpoints to local PSTN circuits, legacy PBXs, and emergency services at that site.

A Cisco IOS XE session border controller that terminates and re-originates SIP signaling and RTP media between independent voice networks, providing topology hiding, protocol interworking, security, and call admission control.

A device or virtual function at a network boundary that controls SIP sessions and media flows, handling security, NAT traversal, protocol normalization, and policy enforcement between dissimilar voice domains.

A SIP element that terminates an incoming dialog and originates a separate outgoing dialog, so it can modify headers, hide topology, and apply policy independently on each call leg.

A logical SIP connection between an enterprise call agent and a service provider or another system that carries multiple concurrent calls over IP instead of physical circuits.

A Cisco IOS configuration object that matches call legs by inbound or outbound direction and destination pattern, then applies the voice port, session target, and settings used to complete the call.

The number string, often containing wildcards such as dots and brackets, that a dial peer uses to match the called number of an outbound call leg.

The process of adding, stripping, or substituting digits in a called or calling number so it matches what the next hop or dial plan expects.

A Cisco Unified Communications Manager configuration that matches dialed digits in a partition and rewrites them before further routing, commonly used to normalize numbers or redirect calls.

A Cisco Unified Communications Manager object that matches dialed digits and points to a gateway, trunk, or route list, optionally applying digit discarding and prefix digits.

An ordered collection of gateways or trunks that a route list selects from, with distribution algorithms that determine which device in the group receives the next call.

An ordered sequence of route groups that Cisco Unified Communications Manager tries in turn, providing failover when devices in earlier groups are unavailable.

A logical grouping of directory numbers and patterns in Cisco Unified Communications Manager that controls which dialable strings a calling search space can reach.

An ordered list of partitions assigned to a device or line that determines which numbers and patterns that originator is allowed to dial.

A Cisco IOS feature that lets a branch router take over call processing for local IP phones when their centralized call agent becomes unreachable, preserving basic internal calling.

A device that converts voice traffic between IP packet networks and traditional circuit-switched telephone networks, handling signaling and media conversion at the boundary.

The algorithm that compresses and decompresses voice or video for transmission; the chosen codec determines bandwidth per call and affects media negotiation between endpoints.

The protocol that carries digitized voice and video streams between endpoints, providing sequence numbers and timestamps so receivers can detect loss and reorder packets.

A Cisco Unified Communications Manager resource that relays media streams between endpoints, used for codec interworking, DTMF conversion, or holding calls during features such as transfer.

The text-based signaling protocol used to establish, modify, and tear down multimedia sessions; it carries call setup messages and negotiates media parameters using SDP.

The format carried inside SIP messages that lists media types, codecs, ports, and addresses so the two ends can agree on how the media streams will be sent.

A SIP call setup approach where the originating side sends its media parameters in the initial INVITE, allowing the terminating side to answer without a separate offer exchange.

A SIP call setup approach where the initial INVITE contains no media description and the answering side supplies the offer, which the originator then answers.

The complete set of rules, patterns, and number normalization logic that determines how a system interprets dialed digits and where it sends each call.

The international numbering standard that expresses phone numbers with a country code and up to fifteen total digits, used to normalize dialed numbers across sites and providers.

A mechanism that limits the number of concurrent calls across a constrained link so voice quality is preserved, rejecting or rerouting calls when bandwidth is exhausted.

The practice of calculating and enforcing the maximum media bandwidth a location or link may consume, typically expressed as the number of allowed calls per codec.

A Cisco Unified Communications Manager service that issues and manages locally significant certificates for endpoints, enabling encrypted signaling and media between phones and the cluster.

Encryption of SIP signaling using Transport Layer Security, protecting call setup messages and credentials from interception between endpoints, gateways, and call agents.

An encrypted form of RTP that protects the media stream itself, providing confidentiality and integrity for voice and video packets in transit.

The techniques, such as static port ranges, application inspection, or a session border controller, that let SIP signaling and RTP media pass through firewalls and NAT devices.

Controls such as calling search space restrictions, class of restriction, and gateway dial-peer permissions that block unauthorized long-distance or international calls from compromised endpoints.

A Cisco IOS gateway mechanism that pairs calling and called restrictions to permit or deny outbound calls on specific dial peers, limiting which callers can reach costly routes.

A Cisco Unified Communications Manager client that collects performance counters, alarms, and traces from cluster nodes for troubleshooting and capacity analysis.

Structured logs written per call that record calling and called numbers, timestamps, duration, and disposition, used for billing, auditing, and troubleshooting.

The collection of actual network traffic on an interface, later analyzed to diagnose signaling errors, one-way audio, or codec mismatches in a failing call.