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CCNA Voice Key Terms
640-461
| Term | Definition |
|---|---|
| analog signal | A method of signaling that uses properties of the transmission medium to convey sound characteristics, such as using electrical properties to convey the characteristics of voice. |
| loop start signaling | A method of signaling that relies on connecting the tip and ring wires connecting to a device to complete an electrical circuit; typically used in devices connecting to the PSTN. |
| ground start signaling | A method of signaling that relies on grounding wires connecting to a device to signal a new call; typically used in PBX systems to avoid glare. |
| glare | An instance in which a user picks up a phone and connects unexpectedly to an incoming call. |
| time-division multiplexing (TDM) | A method of transmitting multiple channels of voice or data over a single digital connection by sending data in specific time slots. |
| channel associated signaling (CAS) | A method of digital signaling in which signaling information is transmitted using the same bandwidth as the voice. |
| common channel signaling (CCS) | A method of signaling in which information is transmitted using a separate, dedicated signaling channel. |
| robbed bit signaling (RBS) | An implementation of channel associated signaling (CAS) that steals the eighth bit of every sixth frame of a digital T1 circuit for signaling information. |
| Q.931 | A signaling protocol used by ISDN CCS implementations. |
| local loop | The PSTN link between the customer premises (such as a home or business) and the telecommunications service provider. |
| private branch exchange (PBX) | A system that allows a company to run an internal, private voice network; PBX systems are usually used in larger companies and provide unique extensions to all devices. |
| key system | A system that allows a company to run a private, internal voice network; key systems are usually used in smaller companies and provide shared-line extensions to all devices, although many key systems now provide unique extensions to all devices. |
| Signaling System 7 (SS7) | The protocol used within the telephony service provider network to provide inter-CO communication and call routing. |
| E.164 | An international numbering plan created by the ITU and adopted for use on the PSTN. |
| quantization | The process of assigning analog signals a numeric value that can be transported over a digital network. |
| Nyquist theorem | Describes the method of converting analog audio signals into digital format by sampling at twice the highest frequency of the audio. |
| mean opinion score (MOS) | A subjective method of determining voice quality; listeners hear a phrase read over a voice network and rate the quality of the audio on a scale of 1 to 5. |
| G.711 | Uncompressed audio codec consuming 64 kbps of bandwidth. |
| G.726 | Compressed audio codec consuming 32 kbps of bandwidth. |
| G.728 | Compressed audio codec consuming 16 kbps of bandwidth. |
| G.729 | Compressed audio codec consuming 8 kbps of bandwidth. |
| Real-time Transport Protocol (RTP) | The UDP-based protocol responsible for transporting audio packets; uses random, even-numbered UDP ports from 16,384 to 32,767 for communication. |
| Real-time Transport Control Protocol (RTCP) | The UDP-based protocol responsible for transporting audio statistics; uses random, odd-numbered UDP ports from 16,384 to 32,767 for communication. |
| Cisco Unified Communications Manager (CUCM) | The multiserver call processing platform that is able to support up to 30,000 users per cluster. |
| Cisco Unified Communications Manager Express (CME) | The call processing platform that is able to support up to 250 users (depending on router hardware). |
| Cisco Unity Connection | The single-server unified messaging platform that is capable of supporting up to 7,500 users. |
| Cisco Unified Presence | Provides status and reachability information for the users of the voice network. |
| Skinny Client Control Protocol (SCCP) | Cisco-proprietary voice-signaling protocol used to control Cisco IP phones. |
| Session Initiation Protocol (SIP) | Voice-signaling protocol created by the IETF as a lightweight alternative to H.323. |
| Internet Telephony Service Provider (ITSP) | Provides VoIP trunk connectivity to the PSTN to provide a cost savings over traditional telephony service providers (TSP). |
| digital signal processors (DSP) | |
| Cisco Unity Express (CUE) | The unified messaging platform that is integrated into a Cisco Unified CME router; capable of supporting up to 250 users. |
| interactive voice response (IVR) | An automated system that provides a recorded, automated process to callers accessing your voice network. |
| Voice Profile for Internet Mail (VPIM) | A feature that allows one voicemail system to exchange messages with another voicemail system. |
| 802.3af Power over Ethernet (PoE) | Industry-standard method of supplying power over an Ethernet cable to attached devices. |
| Cisco Inline Power | Cisco-proprietary, prestandard method of supplying power over an Ethernet cable to attached devices. |
| Cisco Discovery Protocol (CDP) | Protocol that allows Cisco devices to discover other, directly attached Cisco devices. Switches also use CDP to send voice VLAN information to attached IP phones. |
| virtual LAN (VLAN) | A configuration used to break a switch into multiple broadcast domains and IP subnets. |
| trunking | A Cisco-proprietary protocol that replicates VLAN database information to all switches participating in the same VTP domain. |
| 802.1Q | An industry-standard trunking protocol that allows traffic for multiple VLANs to be sent between switches. |
| Dynamic Trunking Protocol (DTP) | Allows switches to dynamically negotiate trunk links. |
| Network Time Protocol (NTP) | Synchronizes the clock of a network device to a more accurate NTP server. |
| Dialed Number Identification Service (DNIS) | Describes dialed number information delivered to a voice processing device. Closely related to Automatic Number Identification (ANI), which identifies caller ID information. |
| Automatic Number Identification (ANI) | Describes caller ID information delivered to a voice-processing device. |
| dial-peer | Logical configuration used to define dial plan information on a Cisco router. |
| Foreign Exchange Station (FXS) ports | Analog interfaces that allow you to connect a legacy analog telephony device to a VoIP network. |
| Foreign Exchange Office (FXO) ports | Analog interfaces that allow you to connect a VoIP network to legacy telephony networks such as the PSTN or a PBX system. |
| Private Line Automatic Ringdown (PLAR) | A configuration used to enable “immediate dial” applications, such as a phone that immediately dials an emergency number when a user lifts the handset. |
| Direct Inward Dial (DID) | A voice configuration that allows users from the PSTN to dial directly into an individual phone at an organization without passing through a receptionist or automated attendant application. |
| Class of Restriction (COR) | The method used to implement calling restrictions in the CME environment. An inbound COR list assigns privileges, whereas an outgoing COR list restricts calling. |
| local directory | The directory that is built automatically by the CME router as you define caller ID information for the ephone-dns. |
| H.450.3 | Industry-standard method of forwarding calls without hairpinning. |
| H.450.2 | Industry-standard method of transferring calls without hairpinning. |
| hairpinning | A problem that occurs when a call is transferred or forwarded from one IP phone to another that keeps the audio path established (or hairpinned) through the original IP phone; this tends to cause QoS issues with the call. |
| call park | A Cisco Unified CME feature that allows you to park a call on hold in a virtual “parking spot” until it can be retrieved. |
| call pickup | A Cisco Unified CME feature that allows you to answer another ringing phone in the network. |
| directed pickup | A method used with call pickup to answer a phone directly by dialing the extension number of the ringing phone. |
| local group pickup | A method used with call pickup to answer a ringing phone from within the local group of an IP phone. |
| other group pickup | A method used with call pickup to answer a ringing phone from another group number, which must be specified after pressing the GPickUp softkey. |
| Single Number Reach | A feature that allows users to have incoming calls ring another, preconfigured number after a defined time limit. Single number reach also allows for mobility, which allows the transfer of an active call to or from the same, preconfigured number. |
| role (CUCM) | Defines a set of privileges to an application’s resources. Privileges may be defined as No Access, Read, or Update. |
| resource | May be an administrative web page, part of a web page, or a tool or interface within an application. |
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