MGCP Configuration Guide, Cisco IOS Release 15M&T
DSP Voice Quality Statistics in DLCX Messages
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DSP Voice Quality Statistics in DLCX Messages

Contents

DSP Voice Quality Statistics in DLCX Messages

The DSP Voice Quality Statistics in DLCX Messages feature provides a way to trace a Media Gateway Control Protocol (MGCP) call between a Cisco PGW 2200 Softswitch and the Cisco IOS gateway by including the MGCP call ID and the DS0 and digital signal processor (DSP) channel ID in call-active and call-history records.

The voice quality statistics are sent as part of the MGCP Delete Connection (DLCX) message. By correlating an MGCP call on the Cisco PGW 2200 Softswitch with a call record on the gateway, you can understand and debug additional statistics from the DSP for problems related to voice quality.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the Feature Information Table at the end of this document. Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to http:/​/​www.cisco.com/​go/​cfn. An account on Cisco.com is not required.

Prerequisites for DSP Voice Quality Statistics in DLCX Messages

You must be using Cisco PGW 2200 version 9.4.1 or a later version with a patch level higher than CSCOgs008/CSCOnn008.

Restrictions for DSP Voice Quality Statistics in DLCX Messages

When the Secure Real-Time Transfer Protocol (SRTP) is enabled, the DLCX message will not report voice quality statistics. The following lines will be omitted for SRTP calls:

  • DSP/Endpoint Configuration (EC)
  • DSP/MOS K-Factor Statistics (KF)
  • DSP/Concealment Statistics (CS)
  • DSP/R-Factor Statistics (RF)
  • DSP/User Concealment (UC)

Information About DSP Voice Quality Statistics in DLCX Messages

Cisco PGW 2200

A call agent (or media gateway controller) and softswitch are industry standard terms used to describe the network element that provides call control functionality to telephony and packet networks. The Cisco PGW 2200 Softswitch functions as a call agent or softswitch in “call control mode.”


Note


All voice quality parameters for Cisco IOS Release 12.4(4)T and later releases are supported only on the Cisco PGW 2200 call agent.


A public switched telephone network (PSTN) gateway provides an interface between traditional Signaling System No. 7 (SS7) or non-SS7 networks and networks based on MGCP, H.323, and Session Initiation Protocol (SIP), which include signaling, call control, and time-division multiplexing/IP (TDM/IP) gateway functions. The Cisco PGW 2200 Softswitch, coupled with Cisco media gateways, functions as a PSTN gateway.


Caution


There is a significant performance degradation on the Cisco PGW 2200 if all connected gateways have the DSP Voice Quality Statistics in DLCX Messages feature enabled. Enabling voice quality statistics on the gateway should only be performed by Cisco personnel.


The Cisco PGW 2200 Softswitch, in either signaling or call control mode, provides a robust, carrier-class interface between PSTN and IP-based networks. Interworking with Cisco media gateways, the Cisco PGW 2200 Softswitch supports a multitude of applications and networks, including the following:

  • Application service provider (ASP) termination
  • Centralized routing and billing for clearinghouse of IP-based networks
  • Dial access
  • International and national transit networks
  • Managed business voice applications
  • Managed voice VPNs
  • Network clearinghouse applications
  • PSTN access for hosted and managed IP telephony
  • PSTN access for voice over broadband networks
  • Residential voice applications

MGCP

MGCP defines the call control relationship between call agents (CAs) and VoIP gateways that translate audio signals to and from the packet network. CAs are responsible for processing the calls.

An MGCP gateway handles the translation between audio signals and the packet network. The gateways interact with a CA, also called a media gateway controller (MGC), which performs signal and call processing on gateway calls. MGCP uses endpoints and connections to construct a call.

Endpoints are sources of or destinations for data and can be physical or logical locations in a device. Connections can be either point-to-point or multipoint. The gateway can be a Cisco router, an access server, or a cable modem, and the CA is a server from a third-party vendor.

Voice Quality Statistics

The Cisco PGW 2200 Softswitch can capture voice quality statistics sent from MGCP-controlled media gateways and can propagate the statistics into call detail records (CDRs) at the end of each call. The Cisco AS5x00 media gateways send voice quality statistics to the Cisco PGW 2200 Softswitch.

Most voice quality statistics are available from the DSP and are controlled using RTP Control Protocol (RTCP) report interval statistics polling. The mean and maximum values are calculated by Cisco IOS software-based polling, which results in additional CPU load for each call. The additional CPU load can be controlled by configuring polling interval by using the ip rtcp report interval command.

The playout delay, playout error, and DSP receive and transmit statistics are automatically polled periodically. Polling for the voice quality statistics, level, and error parameters can be added. For logging the voice quality statistics using syslog, the existing VoIP gateway accounting has been extended. For more information about statistics polling, see the ip rtcp report interval command in the Cisco IOS Voice Command Reference.

Table 1 Voice Quality Statistics for Cisco IOS Release 12.4(4)T and Later Releases

DSP Technology

Platform

Voice Quality Statistics

MSA V6

Cisco AS5350, Cisco AS5350XM, Cisco AS5400, Cisco AS5400HPX, Cisco 5400XM, and Cisco AS5850 with an NPE60 or NPE108 universal port feature card.

DSP/TX

DSP/RX

DSP/PD

DSP/PE

DSP/LE

DSP/ER

DSP/IC

TIC5510

  • Cisco 2800 series and Cisco 3800 series integrated services routers with PVDM2 modules.
  • Cisco VG224 voice gateway
  • Cisco IAD2430 series integrated access devices.
  • Cisco 2600XM, Cisco 2691, Cisco 3700 series access routers, and Cisco 2811, Cisco 2821, Cisco 2851, Cisco 3800 series integrated services routers with the following network modules: NM-HDV2, NM-HDV2-1T1/E1, NM-HD-1V, NM-HD-2V, NM-HD-2VE.
  • Cisco 2821, Cisco 2851, Cisco 3825, and Cisco 3845 with the EVM-HD-8FXS/DID module.

DSP/TX

DSP/RX

DSP/PD

DSP/PE

DSP/LE

DSP/ER

DSP/IC

DSP/EC

DSP/KF

DSP/CS

DSP/RF

DSP/UC

DSP/DL

Quality of Service for Voice

The DSP Voice Quality Statistics in DLCX Messages feature is part of the Cisco quality of service (QoS) technology. QoS is the ability of a network to provide better service to selected network traffic over various technologies, including ATM, Ethernet and 802.1 networks, Frame Relay, SONET, and IP-routed networks that may use any or all of these underlying technologies.

QoS provides the following benefits:

  • Control over bandwidth, equipment, and wide-area facilities—As an example, you can limit the bandwidth consumed over a backbone link by FTP or limit the queueing of an important database access.
  • More efficient use of network resources—Network analysis management and accounting tools enable you to know what your network is being used for and ensure that you are servicing the most important traffic to your business.
  • Ability to customize services—QoS enables ISPs to offer carefully customized grades of service differentiation to their customers.
  • Coexistence of mission-critical applications—Cisco QoS technologies ensure that bandwidth and minimum delays required by time-sensitive multimedia and voice applications are available and that other applications using the link get their fair service without interfering with mission-critical traffic.
  • Foundation for a fully integrated network—Cisco QoS technologies fully integrate a multimedia network, for example, by implementing weighted fair queueing (WFQ) to increase service predictability and by implementing IP precedence signaling to differentiate traffic. In addition, the availability of Resource Reservation Protocol (RSVP) allows you to take advantage of dynamically signaled QoS.

To deliver QoS across a network that comprises heterogeneous technologies (for example, IP, ATM, LAN switches), basic QoS architecture consists of the following three components:

  • QoS within a single network element (for example, queueing, scheduling, and traffic shaping tools).
  • QoS signaling techniques for coordinating end-to-end QoS between network elements.
  • QoS policy, management, and accounting functions to control and administer end-to-end traffic across a network.

Voice Quality Parameters for Cisco IOS Release 12.4(4)T and Later Releases

The following voice quality parameters were introduced in Cisco IOS Release 12.4(4)T:

DSP/EC

The following parameters describe the configuration of a VoIP endpoint. You can define these parameters and they are useful for debugging and logging purposes because they capture the state of the endpoint.

  • CI—Codec ID. A string or a number that identifies the voice codec which is currently used in the call.
  • FM—Frame size. Native frame size, in milliseconds (ms), of the selected codec. An example of a frame size and codec combination is G.729a/30ms. For the G.711 codec, the frame size is a value that you can define in the voice dial peer. For example, G.711 at 80 bytes gives 10 ms per frame. G.711 at 240 bytes gives 30 ms per frame.
  • FP—Frames per packet. Number of codec speech frames encapsulated into a single Real-time Transport Protocol (RTP) packet. Typical values are 1, 2, and 3. Packing lower number of frames per packet results in lower efficiency of IP bandwidth usage. The tradeoff is lower delays and higher robustness of the network.
  • VS—Voice Activity Detection (VAD)-enabled flag. VAD is enabled when VS has a value of one. It results in compression of silent periods leading to reduced or zero packets per second. VAD is disabled when VS has a value of zero. It results in the transmission of continuous packets per second irrespective of active or silent periods on the transmission path.
  • GT—Transmission gain factor (linear). Digital gain multiplier applied to the transmission on the signal path from the PSTN toward the network. GT is applied at the echo canceller Sout port. A gain factor of less than one indicates a loss pad.
  • GR—Reception gain factor (linear). Digital gain multiplier applied to reception on the signal path from the network toward the PSTN. GR is applied at the echo canceller Rin port. A gain factor of less than one indicates a loss pad.
  • JD—Jitter buffer mode. It consists of the following modes:
    • Adaptive mode = 1
    • Fixed mode (no timestamps) = 2
    • Fixed mode (with timestamps) = 3
    • Fixed mode (with passthrough) = 4
  • JN—Jitter buffer nominal playout delay. Size of the jitter buffer in milliseconds. An adaptive jitter buffer tries to make the playout delay equal to the nominal (desired) delay when the observed jitter is small enough to allow this adjustment. For a fixed-mode jitter buffer, the nominal setting is the constant playout delay itself.
  • JM—Minimum playout delay. Minimum playout delay setting for an adaptive-mode jitter buffer. The playout delay never goes below the minimum playout setting even if the observed jitter is zero. This setting is not used for a fixed-mode jitter buffer because the playout delay is fixed and constant at the nominal setting.
  • JX—Maximum playout delay. Sets the limit for increasing the playout delay of an adaptive-mode jitter buffer. An adaptive buffer increases when the jitter is higher than the instantaneous playout delay value.
DSP/KF

K-factor is an endpoint mean opinion score (MOS) estimation algorithm defined in the ITU standard P.VTQ. It is a general estimator of the mean value of a perceptual evaluation of speech quality (PESQ) population for a specific impairment pattern.

The ITU standard P.862 defines and describes the PESQ as an objective method for end-to-end speech quality assessment of narrow band telephone networks and speech codecs.

Mean opinion score (MOS) is associated with the output of a well-designed listening experiment. All MOS experiments use a five-point PESQ scale as defined in the ITU standard P.862.1. The MOS estimate is inversely proportional to frame loss density. Clarity decreases as more frames are lost or discarded at the receiving end.

K-factor represents a weighted estimate of average user annoyance due to distortions caused by effective packet loss such as dropouts and warbles. It does not estimate the impact of delay-related impairments such as echo. It is an estimate of listening quality (MOS-LQO) rather than conversational quality (MOS-CQO), and measurements of average user annoyance range from 1 (poor voice quality) to 5 (very good voice quality).

K-factor is trained or conditioned by speech samples from numerous speech databases, where each training sentence or network condition associated with a P.862.1 value has a duration of eight seconds. For more accurate scores, K-factor estimates are generated for every 8 seconds of active speech.

K-factor and other MOS estimators are considered to be secondary or derived statistics because they warn a network operator of frame loss only after the problem becomes significant. Packet counts, concealment ratios, and concealment second counters are primary statistics because they alert the network operator before network impairment has an audible impact or is visible through MOS.

  • KF—K-factor MOS-LQO estimate (instantaneous). Estimate of the MOS score of the last 8 seconds of speech on the reception signal path. If VAD is active, the MOS calculation is suspended during periods of received silence to avoid inflation of MOS scores for calls with higher silence fractions.
  • AV—Average K-factor score. Running average of scores observed since the beginning of a call.
  • MI—Minimum K-factor score. Minimum score observed since the beginning of a call, and represents the worst sounding 8-second interval.
  • BS—Baseline (maximum) K-factor score. K-factor score that can be obtained for the defined codec.
  • NB—Number of bursts. Number of burst loss events after a call is started. A burst loss is a contiguous run of concealment events of length greater than one.
  • FL—Average frame loss count. Total number of frame losses and concealment events observed after starting a call. The ratio of FL/NB provides the mean burst length in frames. The total concealment duration of the call is provided by the parameter DSP/CS: CT.
  • NW—Number of windows. Total number of K-factor windows observed after starting a call. The number of windows is directly proportional to the duration of a call.
  • VR—Version ID. Version number that identifies a specific K-factor MOS score.
DSP/CS

DSP/CS measures packet (frame) loss and its effect on voice quality in an impaired network. The parameters for concealment statistics are as follows:

  • CR—Concealment ratio (instantaneous). An interval-based average concealment rate, and is the ratio of concealment time over speech time for the last 3 seconds of active speech. When VAD is enabled, calculation of the concealment ratio is suspended during periods of speech silence. During this suspension, it may take more than 3 seconds for a new value to be generated.
  • AV—Average CR. Average of all CR reports after starting a call.
  • MX—Maximum CR. The maximum concealment ratio observed after starting a call.
  • CS—Concealed time. The duration of time in seconds during which some concealment is observed.
  • CT—Total concealment time. The total duration of time in milliseconds during which concealment is observed after starting a call.
  • TT—Total speech time. The duration of time in milliseconds during which active speech is observed after starting a call.
  • OK—Ok time. The duration of time in seconds during which no concealment is observed.
  • SC—Severely concealed time. The duration of time in seconds during which a significant amount of concealment is observed. If the concealment observed is usually greater than 50 milliseconds or approximately five percent, it is possible that the speech is not very audible.
  • TS—Concealment threshold. The threshold in milliseconds used to determine a second as severely concealed. The threshold for concealed seconds is 0 milliseconds, and for severely concealed seconds is 50 milliseconds.
DSP/RF

The R-factor helps in planning voice transmission. In ITU standards G.107 and G.113, the R-factor is defined as follows:

R = Ro - Is - Id - Ie-eff + A

The parameters for the R-Factor are as follows:

  • Ro is based on the signal-to-noise ratio.
  • Is is the simultaneous impairment factor and includes the overall loudness rating.
  • Id is the delay impairment factor and includes talker (Idte) and listener (Idle) echos, and delays (Idd).
  • Ie-eff is the equipment impairment factor and includes packet losses and the types of codecs.
  • A is the advantage factor.
  • ML—R-factor MOS listening quality objective. It reflects only packet loss and codec-related impairments and does not include delay effects.
  • MC—R-factor MOS-CQE.
  • R1—R-factor LQ profile 1.
  • R2—R-factor LQ profile 2.
  • IF—Effective codec impairment (Ie_eff).
  • ID—Idd.
  • IE—Codec baseline score (Ie). The tabulated baseline codec impairment factor.
  • BL—Codec baseline (Bpl). The packet loss robustness factor for the codec being used.
  • R0—R0 (default). The nominal value at which the signal-to-noise ratio is considered nominal.
DSP/UC

The parameters for user concealment are as follows:

  • U1—User concealment seconds 1 count (UCS1)
  • U2—User concealment seconds 2 count (UCS2)
  • T1—UCS1 threshold in milliseconds
  • T2—UCS2 threshold in milliseconds
DSP/DL

The parameters for delay statistics are as follows:

  • RT—Round trip delay
  • ED—End system delay

How to Configure DSP Voice Quality Statistics in DLCX Messages

Configuring DSP Voice Quality Statistics in DLCX Messages

To configure voice quality statistics reporting for MGCP, use the following commands beginning in user EXEC mode.

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    mgcp voice-quality stats [priorityvalue] [all]

    4.    end


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example:
    Router> enable
     

    Enables privileged EXEC mode.

    • Enter your password if prompted.
     
    Step 2 configure terminal


    Example:
    Router# configure terminal
     

    Enters global configuration mode.

     
    Step 3 mgcp voice-quality stats [priorityvalue] [all]


    Example:
    Router(config)# mgcp voice-quality-stats priority 1
     

    Enables voice quality statistics reporting for MGCP.

    The following parameters are sent by default if the priority or all keywords are not used: DSP/TX, DSP/RX, DSP/PD, DSP/PE, DSP/LE, DSP/ER, DSP/IC.

    Priority 1 parameters are: DSP/TX, DSP/RX, DSP/PD, DSP/LE, DSP/EC, DSP/CS, DSP/DL.

    Priority 2 parameters are: DSP/PE, DSP/ER, DSP/IC, DSP/KF, DSP/RF, DSP/UC.

    Using Priority 2 is similar to using the all keyword when the output contains the following parameters: DSP/TX, DSP/RX, DSP/PD, DSP/PE, DSP/LE, DSP/ER, DSP/IC, DSP/EC, DSP/KF, DSP/CS, DSP/RF, DSP/UC, and DSP/DL.

     
    Step 4 end


    Example:
    Router(config)# end
     

    Exits global configuration mode and enters privileged EXEC mode.

     

    What to Do Next

    Use the following Troubleshooting Tips if you did not get the expected results after configuring voice quality statistics reporting for MGCP. See the "Troubleshooting Tips" section for additional guidelines.

    Troubleshooting Tips

    Use the debug mgcp packets command to display statistics reported in the DLCX message generated at the end of a call. The following is sample debug output:

    Router# debug mgcp packets
    
    DLCX 311216 s6/ds1-4/1@as5400a MGCP 0.1
    C: 48A4B
    I: 2
    R: 
    S: 
    X: 4BFAF
    *May 5 10:20:51.643: send_mgcp_msg, MGCP Packet sent to 19.0.2.10:2427 --->
    *May 5 10:20:51.643: 250 311216 OK
    P: PS=1469, OS=28943, PR=1518, OR=29923, PL=0, JI=100, LA=0
    DSP/TX: PK=1448, SG=0, NS=23, DU=206450, VO=39000
    DSP/RX: PK=1449, SG=0, CF=23, RX=206450, VO=38000, BS=0, BP=0, LP=0
    DSP/PD: CU=100, MI=90, MA=110, CO=69352809, IJ=0
    DSP/PE: PC=0, IC=0, SC=0, RM=6, BO=0, EE=0
    DSP/LE: TP=-24, TX=-440, RP=-87, RM=-870, BN=0, ER=50, AC=90, TA=-24, RA=-87
    DSP/ER: RD=0, TD=0, RC=0, TC=0
    DSP/IC: IC=0
    

    Verifying DSP Voice Quality Statistics in DLCX Messages

    Use the following show commands to check your configuration:

    SUMMARY STEPS

      1.    Obtain the call ID by using the show call active voice compact command in privileged EXEC mode.

      2.    Check the status of active calls using the call ID obtained from the show call active voice brief command.

      3.    Verify your configuration using the show call history voice brief command.


    DETAILED STEPS
      Step 1   Obtain the call ID by using the show call active voice compact command in privileged EXEC mode.

      Example:
      Router# show call active voice compact
      
      G<id> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
      Total call-legs: 2
      G11D6 ORG T187 g729r8 TELE P
      G11D6 ORG T0 g729r8 VOIP P 192.0.2.1:19324
      
      Step 2   Check the status of active calls using the call ID obtained from the show call active voice brief command.

      Example:
      Router# show call active voice brief id 11D6
      
      <ID>: <CallID> <start>.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
        dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
       IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
        delay:<last>/<min>/<max>ms <codec>
      
       media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>
      
       long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>
        MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
         last <buf event time>s dur:<Min>/<Max>s
       FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
        <codec> (payload size)
       ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
        <codec> (payload size)
       Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
        MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops>
               speeds(bps): local <rx>/<tx> remote <rx>/<tx>
       Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
       bw: <req>/<act> codec: <audio>/<video>
        tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
       rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
      
      
      Telephony call-legs: 0
      SIP call-legs: 0
      H323 call-legs: 0
      Call agent controlled call-legs: 0
      SCCP call-legs: 0
      Multicast call-legs: 0
      Total call-legs: 0
      
      Step 3   Verify your configuration using the show call history voice brief command.

      Example:
      Router# show call history voice brief
      
      <ID>: <CallID> <start>.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
        dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
       IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
        delay:<last>/<min>/<max>ms <codec>
      
       media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>
      
       long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>
        MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
         last <buf event time>s dur:<Min>/<Max>s
       FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
        <codec> (payload size)
       ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
        <codec> (payload size)
       Telephony <int> (callID) [channel_id] tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
        MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> disc:<cause code>
                   speeds(bps): local <rx>/<tx> remote <rx>/<tx>
       Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
       bw: <req>/<act> codec: <audio>/<video>
        tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
       rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
      
      
      
      Telephony call-legs: 0
      SIP call-legs: 0
      H323 call-legs: 0
      Call agent controlled call-legs: 0
      Total call-legs: 0
      

      Configuration Examples for DSP Voice Quality Statistics in DLCX Messages

      Example: Configuring DSP Voice Quality Statistics in DLCX Messages

      The following example shows how to enable voice quality statistics reporting for MGCP:

      Router> enable
      Router# configure terminal
      Router(config)# mgcp voice-quality-stats
      Router(config)# end

      The following example shows the voice quality parameters selected for priority 1:

      Router(config)# mgcp voice-quality-stats priority 1
      
      16:38:20.461771 192.0.2.1:2427 192.0.2.4:2427 MGCP...... -> 250 1133 OK 
      P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=65, LA=0 
      DSP/TX: PK=118, SG=0, NS=1, DU=28860, VO=2350 
      DSP/RX: PK=0, SG=0, CF=0, RX=28860, VO=0, BS=0, LP=0, BP=0 
      DSP/PD: CU=65, MI=65, MA=65, CO=0, IJ=0 
      DSP/LE: TP=0, RP=0, TM=0, RM=0, BN=0, ER=0, AC=0 
      DSP/IN: CI=0, FM=0, FP =0, VS=0, GT=0, GR=0, JD=0, JN=0, JM=0,
      DSP/CR: CR=0, MN=0, CT=0, TT=0,
      DSP/DC: DC=0,
      DSP/CS: CS=0, SC=0, TS=0,
      DSP/UC: U1=0, U2=0, T1=0, T2=0

      The following example shows the voice quality parameters selected for the all keyword:

      Router(config)# mgcp voice-quality-stats all
      
      16:38:20.461771 192.0.2.1:2427 192.0.2.4:2427 MGCP...... -> 250 1133 OK 
      P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=65, LA=0 
      DSP/TX: PK=118, SG=0, NS=1, DU=28860, VO=2350 
      DSP/RX: PK=0, SG=0, CF=0, RX=28860, VO=0, BS=0, LP=0, BP=0 
      DSP/PD: CU=65, MI=65, MA=65, CO=0, IJ=0 
      DSP/PE: PC=0, IC=0, SC=0, RM=0, BO=0, EE=0 
      DSP/LE: TP=0, RP=0, TM=0, RM=0, BN=0, ER=0, AC=0 
      DSP/ER: RD=0, TD=0, RC=0, TC=0 
      DSP/IC: IC=0
      DSP/EC: CI=0, FM=0, FP =0, VS=0, GT=0, GR=0, JD=0, JN=0, JM=0, JX=0,
      DSP/KF: KF=0, AV=0, MI=0, BS=0, NB=0, FL=0,	
      DSP/CS: CR=0, AV=0, MN=0, MX=0, CS=0, SC=0, TS=0, DC=0,	
      DSP/RF: ML=0, MC=0, R1=0, R2=0, IF=0, ID=0, IE=0, BL=0, R0=0, 
      DSP/UC: U1=0, U2=0, T1=0, T2=0,
      DSP/DL: RT=0, ED=0

      Additional References

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      Cisco IOS Master Commands List, All Releases

      Voice commands

      How to configure QoS for Cisco features

      Cisco IOS Quality of Service Configuration Guide

      Cisco MGC documentation index

      Cisco Media Gateway Controllers

      How to configure MGCP

      Configuring Media Gateway Protocol and Related Protocols

      How to configure QoS for voice applications

      Configuring Quality of Service for Voice

      How to configure voice ports

      Configuring Voice Ports

      Enabling basic management protocols on Cisco access platforms

      Enabling Management Protocols: NTP, SNMP, and Syslog

      Release Notes, Cisco IOS Release 12.3

      Release Notes Index, Cisco IOS Release 12.3

      Standards and RFCs

      Standard/RFC

      Title

      No new or modified standards or RFCs are supported by this feature, and support for existing standards or RFCs has not been modified by this feature.

      MIBs

      MIB

      MIBs Link

      No new or modified MIBs are supported by this feature, and support for existing MIBs has not been modified by this feature.

      To locate and download MIBs for selected platforms, Cisco software releases, and feature sets, use Cisco MIB Locator found at the following URL:

      http:/​/​www.cisco.com/​go/​mibs

      Technical Assistance

      Description

      Link

      The Cisco Support and Documentation website provides online resources to download documentation, software, and tools. Use these resources to install and configure the software and to troubleshoot and resolve technical issues with Cisco products and technologies. Access to most tools on the Cisco Support and Documentation website requires a Cisco.com user ID and password.

      http:/​/​www.cisco.com/​cisco/​web/​support/​index.html

      Feature Information for DLCP Voice Quality Statistics in DLCX Messages

      The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

      Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/​go/​cfn. An account on Cisco.com is not required.

      Table 2 Feature Information for DLCP Voice Quality in DLCX Messages

      Feature Name

      Releases

      Feature Information

      DSP Voice Quality Statistics in DLCX Messages

      12.3(3)

      12.4(4)T

      15.1(3)T

      The DSP Voice Quality Statistics in DLCX Messages feature provides a way to trace a Media Gateway Control Protocol (MGCP) call between a Cisco PGW 2200 and the Cisco IOS gateway by including the MGCP call ID and the DS0 and digital signal processor (DSP) channel ID in call-active and call-history records.

      In Cisco IOS Release 12.4(4)T, new voice quality parameters were introduced.

      The following commands were introduced or modified: debug mgcp, mgcp voice-quality stats.