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What are the parameters of the show call active voice command?

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The show call active voice command is used to troubleshoot echo, jitter, Voice Activity Detection (VAD) and other voice quality problems.

This table provides an explanation for each of the parameters displayed when the command is issued. A number of parameters important for troubleshooting voice quality are highlighted in bold text. Plain Old Telephone Service (POTS) leg parameters are noted beneath a blue heading. VoIP leg parameters are identified beneath a green heading.

Show Call Active Voice Parameter
Explanation of Parameter
GENERIC:Generic stats for the POTS call        leg that follows
SetupTime=866793 msClock time in 100ms increments when the POTS leg was initiated. For incoming ISDN POTS calls, this is the time when the Q931 call setup message was received.
Index=1
PeerAddress=100Destination-Pattern of that matched this POTS peer. For an incoming POTS call leg, this will be the calling number or Automatic Number Identification (ANI).
PeerSubAddress=
PeerId=100Dial peer ID used for this call leg. In this case, (although they do not need to be) the PeerID and the PeerAddress are the same.
PeerIfIndex=9Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.
LogicalIfIndex=5Index used internally to identify the logical interface for the call.
ConnectTime=867030Clock time in in 100ms increments at which the POTS leg was connected. For an incoming ISDN POTS call leg, this is the time when the Q931 call connect message is sent.
CallDuration=00:12:26Time in hh:mm:ss for which the call has been established.
CallState=4Call state for call leg (4=active, 3=connected, 2=connecting). Call state is active.
CallOrigin=2Originate Against the answer (1=originate, 2=answer) for call leg. This gateway answered this (POTS) call leg.
ChargedUnits=0Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.
InfoType=2Information type for this call (1=fax, 2=voice). This is a voice call.
TransmitPackets=37291Number of packets transmitted from Digital Signal Processor (DSP) to telephony interface.
TransmitBytes=725552Byte count equivalent of POTS TransmitPackets value.
ReceivePackets=1689Number of packets received by DSP from telephony interface.
ReceiveBytes=33780Byte count equivalent of POTS ReceivePacketsPackets value.
TELE:POTS call leg
ConnectionId=[0xC59FE183 0xB1700D7 0x0 0x84431C]This is the connection identification number that the gateway has given to uniquely represent this call. It will match across all call legs of the call on this gateway.
TxDuration=746070 msDuration of call(ms) = 12min 26 sec = 746 seconds = 746070 ms
VoiceTxDuration=33780 msCumulative time in ms during which voice packets were sent from the telephony POTS peer to the VoIP gateway. Derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.
FaxTxDuration=0 msCumulative time in ms during which the router was in fax mode.
CoderTypeRate=g729r8Codec used for the call.
NoiseLevel=-59 Active noise level for this call. This value is calculated in the comfort noise generation module and is used for generating comfort noise when VAD is enabled.
ACOMLevel=20Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceler. This value is the sum of the Echo Return Loss (ERL), Echo Return Loss Enhancement (ERLE), and Non-Linear Processing (NLP) loss for the call.
OutSignalLevel=-64Input signal level in Decibels Per Milliwatt (dBm).
InSignalLevel=-58Input signal level in dBm.
InfoActivity=2Active information transfer activity state for this call.
ERLLevel=20Echo Return Loss for this call.
SessionTarget=This value applies to VoIP call legs. This value is specified in the VoIP dial peer. There is no session target for POTS call legs.
ImgPages=0
GENERIC:Generic statistics for VoIP call leg to follow
SetupTime=866928 msClock time in 100ms increments when the VoIP call leg was initiated. For outgoing H323 VoIP calls, this is the time when the H323 call setup message was sent.
Index=1
PeerAddress=200Destination-Pattern of peer. For an outbound VoIP call leg, this will be the called number or Dialed Number Identification Service (DNIS).
PeerSubAddress=
PeerId=200PeerID that DNIS matched. In this case, (although they do not need to be) the peerID and the DNIS are the same.
PeerIfIndex=11
LogicalIfIndex=0
ConnectTime=867029Clock time in in 100ms increments at which the VoIP leg was connected. For an outgoing H323 VoIP call leg, this is the time when the H323 call connect message is received.
CallDuration=00:12:27Duration in hh:mm:ss of call.
CallState=4Call state for call leg (4=active, 3=connected, 2=connecting). Call state is active.
CallOrigin=1Originate against answer (1=originate, 2=answer) for call leg. This gateway originated this (VoIP) call leg.
ChargedUnits=0
InfoType=2
TransmitPackets=1689Number of VoIP packets transmitted by this gateway on this call leg.
TransmitBytes=33780Byte count equivalent of VoIP TransmitPackets value. This should matches VoiceTxDuration from telephony call leg since with G729 1 Byte is sent per 1 ms.
ReceivePackets=37343Number of VoIP packets received by this gateway on this call leg.
ReceiveBytes=746860Byte count equivalent of VoIP ReceivePackets value.
VOIP:VoIP call leg
ConnectionId[0xC59FE183 0xB1700D7 0x0 0x84431C]This is the connection identification number that the gateway has given to uniquely represent this call. It will match across all call legs of the call on this gateway.
RemoteIPAddress=10.1.1.2Remote IP address for the call.
RemoteUDPPort=18280Remote User Datagram Protocol (UDP) port for the call.
RoundTripDelay=53 msRound trip delay as measured by the gateway.
SelectedQoS=best-effortResource Reservation Protocol (RSVP) has not been selected in dial peer for this call.
tx_DtmfRelay=cisco-rtpThe form of Dual-Tone Multifrequency (DTMF) Relay used for the call (if any).
SessionProtocol=cisco

Session Protocol for call - Protocol "cisco" is default for VoIP dial peers, using H.323 signalling and Real Time Protocol (RTP) packets for the voice traffic. Session Initiation Protocol (SIP) is the other VoIP signalling protocol that can be specified  by issuing the session protocol dial peer command. Non-VoIP protocols such as AAL2 for Voice over ATM (VoATM) or Cisco's proprietary Voice over Frame Relay (VoFR) protocol and FRFll for VoFR can also be specified in the appropriate dial peer type.

SessionTarget=ipv4:10.1.1.2Session-target from dial peer. Session target would be ras if a gatekeeper was used.
OnTimeRvPlayout=742740Duration in ms of voice playout from data received on time for this call. The Total Voice Playout Duration can be derived by adding the gap fill durations to the OnTimeRvPlayout duration.
GapFillWithSilence=0 ms

Time(ms) Gateway (GW) played silence. Silence is played out in these situations:

When a packet is lost and there is no audio sample available to play. For example, when two or more packets are lost in sequence. This situation may result in an audible click or gap being heard by the user.

When the playout buffer is adapting to a larger value by inserting silence between buffered voice packets. This situation will not result in an audible loss in quality.

GapFillWithPrediction=0 msDuration in ms of the voice signal played out with signal synthesized from parameters or samples of data preceding it in time. This gap fill occurs because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1 compression algorithms.
GapFillWithInterpolation=0 msAs for GapFillWithPrediction but taking into consideration samples received following the missing voice traffic and stored in the de-jitter buffer. Not currently used.
GapFillWithRedundancy=0 msIf a redundant encoding scheme was used by the transmitter, then the payload of lost or late packets could be partially or fully recovered and played out with a reduced impact on voice quality. This technique is not currently supported.
HiWaterPlayoutDelay=70 msFirst-In, First-Out (FIFO) jitter buffer high mark indicating the maximum depth to which the de-jitter buffer has adapted for this call.
LoWaterPlayoutDelay=69 msFIFO jitter buffer low mark indicating the minimum depth to which the de-jitter buffer has adapted for this call.
ReceiveDelay=69 msCurrent Playout FIFO delay plus decoder delay for the call
LostPackets=0 msLost RTP packets represented in ms. If a packet with an expected RTP sequence number is never received then this value is incremented.
EarlyPackets=1 ms

Number of early RTP packets represented in ms. RTP packets are timestamped as they are transmitted and the RTP timestamp value is included in the packet. The time at which the packet is received is also timed by the gateway's local clock.

If the local clock time difference (time received) of two adjacent packets is smaller than their RTP timestamps difference (time sent) then the second packet is considered to be early.

An early packet can occur when network utilization drops suddenly, resulting in lower network delay for a particular packet.

LatePackets=0 ms

Number of late RTP packets represented in ms. This value is incremented when a packet is received with an RTP sequence number in either of these circumstances:

  • The RTP sequence number is earlier than the RTP sequence number of  the packet currently being played out.
  • The RTP sequence number is later than the currently being played out packet but outside the available playout buffer.
VAD = enabledVAD is enabled for this call leg
CoderTypeRate=g729r8Codec type used for this call
CodecBytes=20Payload size, in bytes, for the codec used
SignalingType=casSignaling type for call - only for permanent calls
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