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Empty RTP stats for successful calls in "show call active voice"

All RTP-relates counters are zero even for successful calls. How can I fix this?

# show call active voice

GENERIC:

SetupTime=120072290 ms

Index=1

PeerAddress=908112937676

PeerSubAddress=

PeerId=76763

PeerIfIndex=71

LogicalIfIndex=0

ConnectTime=120072560 ms

CallDuration=00:03:32 sec

CallState=4

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=10927

TransmitBytes=213612

ReceivePackets=8849

ReceiveBytes=176980

VOIP:

ConnectionId[0x8A2624BE 0xA85F11DF 0xBE24F8CC 0x00000000]

IncomingConnectionId[0x8A2624BE 0xA85F11DF 0xBE24F8CC 0x00000000]

CallID=459869

GlobalCallId=[0x8A26C0E6 0xA85F11DF 0xBE27F8CC 0x00000000]

RemoteIPAddress=192.169.4.17

RemoteUDPPort=15160

RemoteSignallingIPAddress=192.169.4.17

RemoteSignallingPort=5060

RemoteMediaIPAddress=192.169.4.17

RemoteMediaPort=15160

RoundTripDelay=0 ms

SelectedQoS=best-effort

tx_DtmfRelay=rtp-nte

FastConnect=FALSE


AnnexE=FALSE


Separate H245 Connection=FALSE


H245 Tunneling=FALSE


SessionProtocol=sipv2

ProtocolCallId=8A27F936-A85F11DF-BE2AF8CC-00000000@192.169.0.114

SessionTarget=192.169.4.17

OnTimeRvPlayout=0

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=0 ms

LoWaterPlayoutDelay=0 ms

TxPakNumber=0

TxSignalPak=0

TxComfortNoisePak=0

TxDuration=0

TxVoiceDuration=0

RxPakNumber=0

RxSignalPak=0

RxComfortNoisePak=0

RxDuration=0

RxVoiceDuration=0

RxOutOfSeq=0

RxLatePak=0

RxEarlyPak=0

RxBadProtocol=0

PlayDelayCurrent=0

PlayDelayMin=0

PlayDelayMax=0

PlayDelayClockOffset=0

PlayDelayJitter=0

PlayErrPredictive=0

PlayErrInterpolative=0

PlayErrSilence=0

PlayErrBufferOverFlow=0

PlayErrRetroactive=0

PlayErrTalkspurt=0

ClockAdjustDuration=0

EarlyPacketDuration=0

LateDiscardDuration=0

SilenceDiscardDuration=0

JBuffDecreaseDuration=0

JBuffIncreaseDuration=0

OutSignalLevel=0

InSignalLevel=0

LevelTxPowerMean=0

LevelRxPowerMean=0

LevelBgNoise=0

ERLLevel=0

ACOMLevel=0

ErrRxDrop=0

ErrTxDrop=0

ErrTxControl=0

ErrRxControl=0

ReceiveDelay=0 ms

LostPackets=0

EarlyPackets=0

LatePackets=0

SRTP = off

TextRelay = off

VAD = disabled

CoderTypeRate=g729r8

CodecBytes=20

Media Setting=flow-through

CallerName=+999124581487

CallerIDBlocked=False

OriginalCallingNumber=999124581487

OriginalCallingOctet=0x11

OriginalCalledNumber=999112937676

OriginalCalledOctet=0x91

OriginalRedirectCalledNumber=

OriginalRedirectCalledOctet=0x80

TranslatedCallingNumber=999124581487

TranslatedCallingOctet=0x11

TranslatedCalledNumber=999112937676

TranslatedCalledOctet=0x91

TranslatedRedirectCalledNumber=

TranslatedRedirectCalledOctet=0x80

GwReceivedCalledNumber=+999112937676

GwReceivedCalledOctet3=0x0

GwOutpulsedCalledNumber=999112937676

GwOutpulsedCalledOctet3=0x91

GwReceivedCallingNumber=+999124581487

GwReceivedCallingOctet3=0x0

GwReceivedCallingOctet3a=0x80

GwOutpulsedCallingNumber=999124581487

GwOutpulsedCallingOctet3=0x11

GwOutpulsedCallingOctet3a=0x80

MediaInactiveDetected=no

MediaInactiveTimestamp=

MediaControlReceived=

LongDurationCallDetected=no

LongDurCallTimestamp=

LongDurcallDuration=

Username=+999124581487

Appropriate voice dial peer :
        peer type = voice, system default peer = FALSE, information type = voice,
        description = `*** Redirection outgoing LEG ***',
        tag = 76763, destination-pattern = `99112937676',
        voice reg type = 0, corresponding tag = 0,
        allow watch = FALSE
        answer-address = `', preference=0,
        CLID Restriction = None
        CLID Network Number = `'
        CLID Second Number sent
        CLID Override RDNIS = disabled,
        rtp-ssrc mux = system
        source carrier-id = `', target carrier-id = `',
        source trunk-group-label = `',  target trunk-group-label = `',
        numbering Type = `unknown'
        group = 76763, Admin state is up, Operation state is up,
        incoming called-number = `', connections/maximum = 3/unlimited,
        DTMF Relay = enabled,
        modem transport = system,
        URI classes:
            Incoming (Request) =
            Incoming (To) =
            Incoming (From) =
            Destination =
        huntstop = disabled,
        in bound application associated: 'DEFAULT'
        out bound application associated: ''
        dnis-map =
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        Translation profile (Incoming):
        Translation profile (Outgoing):
        incoming call blocking:
        translation-profile = `'
        disconnect-cause = `no-service'
        advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
        mailbox selection policy: none
        type = voip, session-target = `ipv4:192.168.4.17',
        technology prefix:
        settle-call = disabled
        ip media DSCP = ef, ip media rsvp-pass DSCP = ef
        ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
        ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
        ip video rsvp-fail DSCP = af41,
        ip defending Priority = 0, ip preemption priority = 0
        ip policy locator voice:
        ip policy locator video:
        UDP checksum = disabled,
        session-protocol = sipv2, session-transport = system,
        req-qos = best-effort, acc-qos = best-effort,
        req-qos video = best-effort, acc-qos video = best-effort,
        req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
        req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,
        dtmf-relay = rtp-nte,
        RTP dynamic payload type values: NTE = 101
        Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
               CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
               A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, lmr_tone=0, nte_tone=0
               h263+=118, h264=119
               G726r16 using static payload
               G726r24 using static payload
        RTP comfort noise payload type = 19
        fax rate = voice,   payload size =  20 bytes
        fax protocol = system
        fax-relay ecm enable
        Fax Relay ans enabled
        Fax Relay SG3-to-G3 Enabled (by system configuration)
        fax NSF = 0xAD0051 (default)
        codec = transparent,   payload size =  0 bytes,
        video codec = None
        voice class codec = `'
        voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30
        voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
        voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30
        voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
        text relay = disabled
        Media Setting = flow-through (voice class media 1)
        Expect factor = 10, Icpif = 20,
        Playout Mode is set to adaptive,
        Initial 60 ms, Max 1000 ms
        Playout-delay Minimum mode is set to default, value 40 ms
        Fax nominal 300 ms
        Max Redirects = 1, signaling-type = cas,
        VAD = disabled, Poor QOV Trap = disabled,
        Source Interface = NONE
        voice class sip url = system,
        voice class sip rel1xx = system,
        tvoice class sip outbound-proxy = system,
        voice class sip asserted-id = system,
        voice class sip privacy = system,
        voice class sip e911 = system,
        voice class sip history-info = system,
        voice class sip anat = system,
        voice class sip g729 annexb-all = system,
        voice class sip early-offer forced = system,
        voice class sip negotiate cisco = system,
        redirect ip2ip = disabled
        local peer = false
        probe disabled,
        Secure RTP: system (use the global setting)
        voice class perm tag = `'
        Time elapsed since last clearing of voice call statistics never
        Connect Time = 22557291, Charged Units = 0,
        Successful Calls = 1739, Failed Calls = 3, Incomplete Calls = 0
        Accepted Calls = 0, Refused Calls = 0,
        Last Disconnect Cause is "10  ",
        Last Disconnect Text is "normal call clearing (16)",
        Last Setup Time = 11404330.
        Last Disconnect Time = 11404377.

Everyone's tags (3)
1 REPLY
Cisco Employee

Re: Empty RTP stats for successful calls in "show call active vo

What type of call is the output for? What's the call flow? How is the gateway configured?

-Felipe

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