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ACOM
The total echo return loss on the network. ACOM measures how significantly the voice gateway reduced the echo. ACOM includes echo reduction that occurs with or without the activity of an echo cancellation device.
all channels busy
The percentage of the reporting interval for which all active channels in a trunk group carried traffic.
analog telephone adapters (ATA)
A device used to connect a standard telephone to a computer or network so that the user can make calls over the internet. ATAs are typically cheaper than specialized VoIP phones that connect directly to a computer's USB port. An ATA typically supports one or two ports.
Answer Seizure Ratio (ASR)
The number of successfully answered calls compared to the number of call attempts.
Application Enablement Services (AES)
The Avaya application server that provides system management APIs.
audio/visual conferencing server
A server in a Microsoft Lync environment that enables audio and video (multi-party) conference calls. Also referred to as an A/V MCU.
Automatic Number Identification (ANI)
A feature of telephony that lets subscribers display or capture the telephone numbers of calling parties.
burst
The points in a data stream when a high percentage of packets is lost or discarded due to packets arriving late.
burst density
The percentage of packets within burst periods that are lost or discarded.
burst duration
The average duration of all high-loss periods in a data stream.
busy-hour call attempts (BHCA)
The number of calls attempted at the busiest (peak) hour of the day.
busy-hour call completions (BHCC)
The number of calls completed at the busiest (peak) hour of the day. BHCC is a measure of the throughput capacity of a VoIP network.
call detail record (CDR)
Storage of information about the endpoints of a call and other aspects of call control and routing.
call leg
A discrete segment of a call connection in a VOIP network. A logical connection between a router and an endpoint.
call management record (CMR)
Storage of information about the quality of the streamed audio of a call.
call minutes
The number of minutes that calls were active during the selected time period.
call path
The path, or route, a call takes between the origination and destination endpoints in a network.
call setup
A series of connections that occur between a telephone placing a VoIP call and the active call server. The call server is responsible for certain signaling to the telephone that allows it to play a dial tone and make the call. The call server also establishes a connection to the destination endpoint in the PSTN. The call setup protocol defines the messages that are passed among the call server, gateway, and endpoints.
call setup failures
The calls that fail to connect during the setup phase. Expressed as a percentage of all calls that were attempted during the monitoring interval.
call setup protocol
Protocols involved in the call setup process: SIP, SCCP, H323, and MGCP.
calls attempted
All calls that the monitored system tried to place, either successfully or unsuccessfully. This metric is the primary unit of measurement for the Call Volume Audio views.
calls completed
The number of audio-only calls that were successfully completed during the selected time frame. Includes calls from endpoints within the monitored system and calls from outside of the system to endpoints inside of the system.
channels out-of-service
The number of channels in a trunk group that are out of service.
Cisco CallManager cluster
A group of physical servers, running Cisco Unified Communications Manager (CallManager), to work together as an IP PBX system.
Cisco IP Communicator
A Microsoft Windows-based softphone application for making voice and video calls.
Cisco Performance Monitor
A feature of Cisco routers and switches that enables reporting of quality metrics for a medianet environment.
codec
Codecs (the term is short for coder-decoder) convert an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay.
concealment
A technique for masking the effects of packet loss in VoIP communications. Also known as packet loss concealment (PLC).
concealment ratio
The percentage of frames in a data stream that are concealment frames, which the endpoints generate to conceal packet loss. Includes both early and late packets.
conference ID
Identifier for a voice gateway call.
connection attempts
The number of times a connection to the server is attempted before timing out.
controller LAN board (C-LAN)
G650 voice gateways can have C-LANs defined and running on the device as separate call servers. Each C-LAN has a dedicated IP address, which appears in UC Monitor reports as a call server. However, the actual call server is the Communication Manager, which is usually installed on a separate media server.
conversational MOS
The Mean Opinion Score (MOS) based on metric factors from both directions of data flow.
currently missing phones
The percentage of endpoints that were registered to a server in the group, but are no longer registered to any server in the group, and were never observed to have been formally unregistered.
delay
see latency
delay to dial tone
The amount of time it takes for a user to hear a dial tone after picking up the receiver of an IP telephone. During the call setup phase of a VoIP call, the device receives messages from the call server to play a dial tone. Users can think that the system is not working when dial tone is delayed.
Differentiated Services Code Point (DSCP)
The Differentiated Services Code Point setting of the incoming RTP packets.
digital telephone
Digital telephones convert analog sound into digital format at the handset. Digital telephones do not include web browsers or more advanced applications generally available from IP telephones.
directory number (DN)
A telephone number.
echo
The phenomenon of your voice coming back to you, as if you were repeating yourself. In a VoIP network, echo is accentuated by the amount of delay in the network.
Echo Return Loss (ERL)
Reduction in the echo level produced in the circuit without an echo canceler. The degree or amount of loss reflects the volume of the echo that remains, and a measurement of how significantly echo was reduced.
Echo Return Loss Enhancement (ERLE)
An enhancement in the echo return loss that an echo canceler produces. An echo canceler removes the echo portion of a VoIP call signal as it exits the tail circuit and heads into the WAN. Also referred to as cancellation loss.
echo tail length
The “length” of echo cancellation processing. Based on the distance between a voice gateway and the endpoint. Typical values range from 8 milliseconds to 32 milliseconds.
edge server
In a Microsoft Lync environment, a server running in the perimeter network to provide connectivity for external users and public instant messaging connections. The edge server ensures that users outside the firewall are authorized before they obtain access to the Lync deployment. The edge server also provides media relay for audio/visual streams where direct connection is not possible.
egress interface
The interface where traffic exits a device.
endpoint
An endpoint is any device where a media stream begins or ends, such as telephone, softphone, telepresence, voice gateway, media device, and video camera.
erlang
In telephony, a statistical measure of the volume of telecommunications traffic. Traffic of one Erlang refers to a single resource being in continuous use, or two channels being at fifty percent use, and so on.
failover
Failover is the process of switching to a backup server or system when the primary server or system fails, is offline, or becomes unavailable.
Flexible NetFlow
The next generation in flow technology from Cisco. Flexible NetFlow enables the delivery of medianet data to UC Monitor.
front-end server
In a Microsoft Lync environment, a server that typically performs call processing functions. Microsoft Lync supports a pool of one or more front-end servers working together to perform functions such as call processing.
frozen period
The average length of frozen video instances.
frozen video frequency
The frequency of long and noticeable frozen video periods for an entire session. Expressed as a percentage of session time.
G.107
An ITU-T standard for reporting VoIP conversational call quality. UC Monitor uses this standard to calculate MOS from the voice gateway’s perspective at the end of an IP-PSTN call.
G.711
A high-performance, high bit-rate codec (64 Kbps) often used for its excellent voice quality. Because it does not use compression, G.711 requires more bandwidth than some other common codecs.
gap density
The percentage of lost or discarded packets in the gaps between bursts in a data stream.
gap duration
The average duration of periods of good performance (low loss) between periods of data loss in a data stream.
gatekeeper
An optional component of a VoIP network that provides services such as endpoint registration, address resolution, admission control, and user authentication.
gateway
A device that provides the conversion interface between the PSTN and an IP network.
Grade of Service (GoS)
An estimation of the probability that a VoIP call receives a busy signal. The GoS value (a decimal fraction) is always expressed with reference to the busy hour when the traffic intensity is the greatest. GoS is reported from the perspective of the origination Location or gateway device (the outgoing direction).
group overflows
The number of outgoing calls presented to the trunk but not carried. Overflow calls arrived when all trunks in the trunk group were busy, but were not queued on the trunk group. This value does not include calls that were denied service on the trunk group because of authorization failures.
H.323
An ITU standard protocol for call setup. UC Monitor supports gateways that use this protocol to communicate with Cisco Unified Communications Manager.
ingress interface
The interface where traffic enters a device.
Interactive Connectivity Establishment (ICE)
A mechanism for SIP-based VoIP clients to successfully traverse the variety of firewalls that may exist between a remote user and a network.
jitter
Packet delay that distorts the quality of a voice conversation.
jitter buffer
Buffers that attempt to reduce or eliminate network jitter by caching packets. If jitter exceeds caching capacity, packets are lost (jitter buffer loss).
jitter buffer delay
Delay that the jitter buffer introduces while it holds one or more packets to reduce variations in packet arrival times.
jitter buffer loss
The packets that are lost when jitter hinders the caching capacity of the jitter buffer.
jitter buffer over runs
The number of times that jitter exceeded the maximum size setting of the jitter buffer. Packets arrive too quickly to be contained by the jitter buffer. Over runs usually result in packet loss.
jitter buffer under runs
The number of times that the jitter buffer became empty. Packets arrive too slowly to be contained by the jitter buffer. Under runs usually indicate that delays are too lengthy for the buffer setting.
keepalive
A message sent by one device to another to verify that the connection between the two is operating, or to prevent the connection from breaking.
latency
One-way delay. Calculated from the origination party to the destination party. Includes propagation delay, network delay, and packetization delay.
Listening MOS
The Mean Opinion Score, which is based on call legs traveling toward the endpoint to reflect listener perception of quality.
Mean Opinion Score (MOS)
The Mean Opinion Score (MOS) is an industry standard method for gauging call quality. MOS is an estimation of how impairments to a voice signal affect listener perception of call quality.
mean time to repair (MTTR)
Time required to repair a failed component or device. MTTR is also defined as "mean time to recovery," which is the amount of time required for a device to recover from a failure.
media device
Specialized devices to route calls from the PSTN, handle conference calls, or transcode media streams. Examples include voice gateways, mediation servers, conferencing servers, and unified messaging servers.
Media Gateway Control Protocol (MGCP)
Signaling and call control protocol used in a distributed VoIP system.
media processor
The IP termination point for audio. It performs the conversion between time-division multiplexing (TDM) and IP. The audio payload is encapsulated in RTP, then UDP, then IP.
media relay
An edge server function used with interactive connectivity establishment to provide end-to-end delivery of media streams where direct connectivity between two IP endpoints is not possible.
medianet
A medianet is an IP architecture that enhances the performance of video, voice, and data, and automates many aspects of configuration.
mediation server
Handles calls from the PSTN and interoperates with media devices that are outside the Microsoft Lync environment, such as other IP telephony environments.
midstream device
A medianet-enabled device, such as a router or switch, that sends NetFlow data to UC Monitor to report on the quality of audio or video streams.
narrowband codec
Compresses and decompresses traditional speech, covering frequencies 300 to 3400 Hz, to more easily fit over an IP network.
NetFlow
Developed by Cisco, this network protocol collects IP traffic information.
network congestion
Occurs when a network device carries so much data that the QoS deteriorates. Network congestion can result in packet loss and delay.
network delay
Transport delay produced by intervening network equipment, such as routers and switches.
Network MOS
MOS listening quality value that is based only on network factors, such as codec, packet loss, packet reordering, packet errors, and jitter.
noise level
(Microsoft only) The average portion of an audio signal that is noise and not actual voice data. Measured in decibels.
origination/destination
The origination phone initiates the call. The destination phone receives the call.
packet loss
The percentage of data packets that were lost in transit. These packets were sent but never received at the destination.
packet rate
The number of data packets that are received per second. UC Monitor uses this value to determine whether an RTP stream is audio or video for medianet-enabled devices that do not report a codec.
packetization delay
Delay introduced by a codec.
port mirroring
On a network switch, the port mirroring function sends copies of network packets from one port to another switch or port for analysis. The port mirroring function on Cisco switches is named Switched Port Analyzer (SPAN).
post-dial delay
The amount of time from when a user enters the last digit of a telephone number to when the user hears a ring or busy signal.
POTS
Plain Old Telephone System. The voice-grade telephone service that is the basic form of residential and small business service connection to the telephone network in most parts of the world.
presence
In a unified communications environment, the ability for users to know the status and availability of other users.
propagation delay
Delay produced by the physical distance that packets travel in a data transmission.
PSTN
Public Switched Telephone Network. The network of the world's public circuit-switched telephone networks, in much the same way that the internet is the network of the world’s public IP-based packet-switched networks.
Publisher
Required member of the Cisco Unified Communications Manager cluster that publishes database (config) updates to other members of the cluster. In failover situations, the Publisher can take over call processing functions from the Subscriber.
Quality of Service (QoS)
QoS provides different priorities, or throughput levels, for different applications, users, or data flows on a packet-switched telecommunications network.
Quality Report Tool (QRT)
A problem-reporting tool for Cisco IP phones, which allows users to easily report audio and other general problems with their IP phone. Many Cisco phones have a QRT softkey.
queue abandons
Calls that were removed from the Trunk Group Queue.
queue overflows
The number of calls that arrived when all slots in the Trunk Group Queue were busy.
Real-Time Control Protocol
An IETF standard for providing out-of-band statistics and control information for an RTP flow. Generally sent over the next highest odd-numbered port as the corresponding RTP flow. RTCP provides feedback on the QoS in media distribution.
Real-Time Transport Protocol (RTP)
An IETF standard for delivering audio and video over the Internet. Generally sent on an even-numbered UDP port.
R-value
R-value is a number, or score, that is used to quantitatively express the subjective quality of speech in a VoIP network. The R-value can range from 1 (worst) to 100 (best), and is based on the percentage of users who are satisfied with the quality of a test voice signal after it passed through a network from a source (transmitter) to a destination (receiver). In many cases, an R-value is mapped to a MOS, which is used most frequently when referring to VoIP call quality.
sending/receiving
All endpoints that are involved in a call are, at some point, a sender and a receiver. During a call, all endpoints send data (talk) and receive data (listen).
sequence falls
The number of times that at least one packet arrived out of order.
sequence jumps
The number of times that at least one consecutive packet was lost.
Session Initiation Protocol (SIP)
A signaling protocol for setting up and tearing down multimedia communication sessions such as voice and video calls.
severely concealed seconds
The number of call seconds that had more than 5 percent concealment events from the start of the data stream.
signal level
The average audio signal level in decibels (dBm0).
Simple Network Management Protocol (SNMP)
Protocol for managing devices on IP networks. A network that is managed by SNMP consists of a managed device, an agent on the managed device, and a network management system on the manager.
SIP Enablement Services
The SIP proxy server for Avaya SIP endpoints.
SIP trunking
A service offered by an Internet Telephony Service Provider that permits businesses with a PBX to use VoIP outside the enterprise network by using the same connection as the Internet connection.
Skinny Call Control Protocol (SCCP)
A proprietary Cisco messaging protocol that is used between clients (phones) and the Cisco Unified Communications Manager in a VoIP environment. SCCP passes messages using TCP and port 2000.
stream leg
Unidirectional stream of packets.
Subscriber
A server in the Cisco Unified Communications Manager cluster that typically performs call-processing functions.
successful call ratio
Number of successful call completions divided by the number of call attempts.
Time to Live (TTL)
A counter embedded in data to prevent a data packet from circulating through the network indefinitely. The counter decrements each time that the packet passes through a router or a switch.
traceroute
A diagnostic tool that displays the route (path) and measures the transit delays of packets across an IP network
trunk group
A group of trunks serving the same special purpose. The term commonly is applied to voice Private Branch Exchange (PBX) trunks.
trunk group ID
The trunk group number for a voice gateway call in a Call Watch. The phone number of the endpoint that sends data through the gateway cannot be identified while the call is in progress. The trunk group ID is substituted for the phone number and displayed in the Phone Number field until the phone number is identified.
unified communications
The convergence of multiple modes of communication (such as phone, video, and email) within applications and infrastructure to allow people, teams, and organizations to communicate more effectively.
Uniform Resource Identifier (URI)
A user’s SIP phone number. A SIP URI may resemble an email address, such as sip:john.smith@ca.com.
VG 224 gateway
A Cisco gateway device that lets analog phones connect to an IP PBX, which typically supports 24 analog phones. All phones are assigned the same IP address with different port numbers.
video bit rate
The number of bits sent per second for an entire video stream. Bit rates provide a gauge of codec performance.
video frame decoding time
The average amount of time for decoding frames in a stream. A slower decoding rate can be the result of conditions on the endpoint, such as lack of CPU resources, and can affect call quality.
video frame loss
The average number of unique consecutive images, or video frames, lost due to corruption and error concealment for the entire system. Video frames can span multiple packets.
video frame rate
The average number of frames that were sent or received per second for an entire stream.
video jitter
The variation in delay among video packets in the same stream.
video latency
The maximum time for a video packet to travel between the calling parties. Measured from end-to-end in one direction. Calculated by taking the average round-trip time for a call leg in a given video call and dividing it in half.
video packet loss
The percentage of video packets that were lost in transit. These packets were sent but never received at the destination.
VLAN ID
The ID of the virtual local area network (VLAN) that carries RTP packets.
voice gateway
A router or switch with a specialized card that enables VoIP calls to and from the PSTN.
Voice over IP (Voip)
A set of technologies, protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.
VoIP trunk
A large bandwidth channel that handles multimedia data and forms the backbone of a network. In telephone exchanges, a trunk simultaneously transmits data and voice packets from one point to another.
weighted average
A computation for an average value that takes into consideration the number of observations.
For example, to compute the value of average jitter across sites in your network, use the following calculation:
(site1 avg + site2 avg + site3 avg + site N avg) divided by N
where N is the number of sites.
To compute weighted average, use the following calculation, which allots more weight to sites with more jitter:
(o1(s1) + o2(s2) + o2(s3)+ oN(sN)) divided by (o1 + o2 + o3)
where N is the number of sites and o is the number of observations.
wideband codec
Compresses and decompresses wideband speech, such as high-definition voice, to more easily fit over an IP network.
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