Table Of Contents
DSP Voice-Quality Statistics in DLCX Messages
Prerequisites for DSP Voice-Quality Statistics in DLCX Messages
Information About DSP Voice-Quality Statistics in DLCX Messages
Voice Quality Parameters for Cisco IOS Release 12.4(4)T
How to Configure DSP Voice-Quality Statistics in DLCX Messages
Configuring DSP Voice-Quality Statistics in DLCX Messages
Verifying DSP Voice-Quality Statistics in DLCX Messages
Troubleshooting DSP Voice-Quality Statistics in DLCX Messages
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 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.
These voice quality statistics are sent as part of the MGCP Delete Connection (DLCX) message. By correlating an MGCP call on the Cisco PGW 2200 with the call record on the gateway, additional statistics from the DSP can be understood and debugged for problems related to voice quality.
Release Modification12.3(3)
This feature was introduced.
12.4(4)T
Introduced new voice quality parameters and modified the mgcp voice-quality stats command.
Feature History for DSP Voice-Quality Statistics in DLCX Messages
Finding Support Information for Platforms and Cisco IOS Software Images
Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.
Contents
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Prerequisites for DSP Voice-Quality Statistics in DLCX Messages
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Information About DSP Voice-Quality Statistics in DLCX Messages
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How to Configure DSP Voice-Quality Statistics in DLCX Messages
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.
Information About DSP Voice-Quality Statistics in DLCX Messages
To configure the DSP Voice-Quality Statistics in DLCX Messages feature, you should understand the following concepts:
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MGCP
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 in "call control mode" functions as a call agent or softswitch.
Note
All voice quality parameters upto Cisco IOS Release 12.4(4)T are supported only on the Cisco PGW 2200 call agent.
A PSTN gateway provides the interface between traditional SS7 networks or non-SS7 networks and networks based on Media Gateway Control Protocol (MGCP), H.323, and Session Initiation Protocol (SIP), including signaling, call control, and time-division multiplexing/IP (TDM/IP) gateway functions. The Cisco PGW 2200, coupled with Cisco media gateways, functions as a PSTN gateway.
CautionThere is a significant performance degradation on the Cisco PGW 2200 if all connected gateways have this feature enabled.
Enabling voice quality statistics on the gateway should only be performed by Cisco personnel.
The Cisco PGW 2200, in either signaling mode or call control mode, provides a robust, carrier-class interface between the PSTN and IP-based networks. Interworking with Cisco media gateways, the Cisco PGW 2200 supports a multitude of applications, including the following:
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International and national transit networks
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Dial access
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Application service provider (ASP) termination
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Managed business voice applications
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Managed voice virtual private networks (VPNs)
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PSTN access for hosted and managed IP telephony
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Residential voice applications
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PSTN access for voice over broadband networks
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Network clearinghouse applications
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Centralized routing and billing for clearinghouse of IP-based networks
MGCP
MGCP defines the call control relationship between call agents (CAs) and VoIP gateways that translate audio signals to and from the packet network. The 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, access server, or cable modem, and the CA is a server from a third-party vendor.
Voice Quality Statistics
The Cisco PGW 2200 can capture voice quality statistics sent from MGCP-controlled media gateways and can propagate the statistics into the call detail records (CDRs) at the end of each call. The Cisco AS5x00 media gateways send voice quality statistics to the Cisco PGW 2200.
Most voice quality statistics are available from the DSP and are controlled with RTP Control Protocol (RTCP) report interval statistics polling. The mean and maximum values are calculated by Cisco IOS software-based polling. This results in additional CPU load for each call. The additional CPU load can be controlled by the configured polling interval using the ip rtcp report interval commands.
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. Use the ip rtcp report interval command reference for more information about statistics polling.
Quality of Service for Voice
The DSP Voice-Quality Statistics in DLCX Messages feature is part of Cisco quality of service (QoS) technologies. QoS is the ability of a network to provide better service to selected network traffic over various technologies, including Frame Relay, ATM, Ethernet and 802.1 networks, and SONET, as well as IP-routed networks that may use any or all of these underlying technologies.
QoS provides the following benefits:
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Control over bandwidth, equipment, and wide-area facilities—As an example, you can limit the bandwidth consumed over a backbone link by FTP or queueing of an important database access.
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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.
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Customized services—QoS enables Internet service providers (ISPs) to offer carefully customized grades of service differentiation to their customers.
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Coexistence of mission-critical applications—Cisco QoS technologies make certain 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.
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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 IP precedence signaling to differentiate traffic. Also available is Resource Reservation Protocol (RSVP), which 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), the basic QoS architecture has three components:
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QoS within a single network element (for example, queueing, scheduling, and traffic shaping tools).
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QoS signaling techniques for coordinating end-to-end QoS between network elements.
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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
Cisco IOS Release 12.4(4)T introduces these new voice quality parameters:
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DSP/EC : Endpoint Configuration
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DSP/KF : MOS/K-Factor Statistics
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DSP/CS: Concealment Statistics
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DSP/UC: User Concealment Statistics
DSP/EC : Endpoint Configuration
The following elements describe the configuration of the VoIP endpoint. They are provisioned by the user and are used to debug and log, because they capture the state of the endpoint.
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CI—Codec ID
A string or number that identifies the voice codec which is currently used in the call.
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FM—Frame size in milliseconds
Native frame size 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 is provisioned by the user in the voice dial peer. For example, G.711 at 80 bytes gives 10 milliseconds per frame. G.711 at 240 bytes gives 30 milliseconds per frame.
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FP—Frames per packet
Number of codec speech frames encapsulated into a single RTP packet. Typical values are 1, 2, and 3. Packing lower frames per packet results in lower efficiency of IP bandwidth usage. The tradeoff is lower delays and higher robustness of the network.
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VS—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.
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GT—Transmission gain factor (linear)
Digital gain multiplier applied to transmission on the signal path from the PSTN toward the network. It is applied at the echo canceller Sout port. A gain factor less than one indicates a loss pad.
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GR—Reception gain factor (linear)
Digital gain multiplier applied to reception on the signal path from the network toward the PSTN, applied at the echo canceller Rin port. A gain factor less than one indicates a loss pad.
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JD—Jitter buffer mode
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Adaptive mode = 1
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Fixed mode (no timestamps) = 2
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Fixed mode (with timestamps) = 3
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Fixed mode (with passthrough) = 4
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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.
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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.
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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 : MOS/K-Factor Statistics
K-factor is an endpoint MOS estimation algorithm defined in ITU standard P.VTQ. It is a general estimator and is used to estimate the mean value of a PESQ population for a specific impairment pattern.
ITU standard P.862 defines and describes the perceptual evaluation of speech quality (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 a term that relates to the output of a well designed listening experiment. All MOS experiments use a five point PESQ scale as defined in ITU standard P.862.1. The MOS estimate is a number 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 eight 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.
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KF—k-factor MOS-LQO estimate (instantaneous)
Estimate of the MOS score of the last eight 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.
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AV—Average k-factor score
Running average of scores observed since the beginning of a call.
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MI—Minimum k-factor score
Minimum score observed since the beginning of a call, and represents the worst sounding eight second interval.
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BS—Baseline (maximum) k-factor score
K-factor score that can be obtained for the provisioned codec.
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NB—Number of bursts
Number of burst loss events after starting a call. A burst loss is a contiguous run of concealment events of length greater than one.
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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 in the parameter DSP/CS: CT.
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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.
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VR—Version ID
Version number that identifies a specific k-factor MOS score.
DSP/CS: Concealment Statistics
It measures packet (frame) loss and its effect on voice quality in an impaired network.
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CR—Concealment ratio (instantaneous)
An interval-based average concealment rate, and is the ratio of concealment time over speech time for the last three 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 three seconds for a new value to be generated.
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AV—Average CR
Average of all CR reports after starting a call.
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MX—Maximum CR
The maximum concealment ratio observed after starting a call.
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CS—Concealed seconds
The duration of time during which some concealment is observed.
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CT—Total concealment time in milliseconds
The total duration of time during which concealment is observed after starting a call.
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TT—Total speech time in milliseconds
The duration of time during which active speech is observed after starting a call.
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OK—Ok seconds
The duration of time in seconds during which no concealment is observed.
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SC—Severely concealed seconds
The duration of time during which a significant amount of concealment is observed. If the concealment observed is usually greater than fifty milliseconds or approximately five percent, it is possible that the speech is not very audible.
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TS—Concealment threshold in milliseconds (ms)
The threshold used to determine a second as severely concealed. The threshold for concealed seconds is 0 ms, and for severely concealed seconds is 50 ms.
DSP/RF—R-Factor Statistics
The R-factor helps in planning voice transmission. In ITU standards G.107 and G.113, the R-factor is defined as:
R = Ro - Is - Id - Ie-eff + A
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Ro is based on the signal to noise ratio.
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Is is the simultaneous impairment factor and includes the overall loudness rating.
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Id is the delay impairment factor and includes talker (Idte) and listener (Idle) echos, and delays (Idd).
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Ie-eff is the equipment impairment factor and includes packet losses and the types of codecs.
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A is the advantage factor.
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ML—R-factor MOS listening quality objective
Reflects only packet-loss and codec related impairments and does not include delay effects.
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MC—R-factor MOS-CQE
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R1—R-factor LQ profile 1
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R2—R-factor LQ profile 2
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IF—Effective codec impairment (Ie_eff)
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ID—Idd
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IE—Codec baseline score (Ie)
The tabulated baseline codec impairment factor.
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BL—Codec baseline (Bpl)
The packet loss robustness factor for the codec being used.
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R0—R0 (default)
The nominal value at which the signal-to-noise ratio is considered nominal.
DSP/UC: User Concealment Statistics
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U1—User concealment seconds 1 count (UCS1)
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U2—User concealment seconds 2 count (UCS2)
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T1—UCS1 threshold in ms
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T2—UCS2 threshold in ms
DSP/DL: Delay Statistics
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RT—Round trip delay
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ED—End system delay
How to Configure DSP Voice-Quality Statistics in DLCX Messages
This section contains procedures for configuring the DSP Voice-Quality Statistics in DLCX Messages feature. Each procedure is identified as either required or optional.
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Configuring DSP Voice-Quality Statistics in DLCX Messages (required)
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Verifying DSP Voice-Quality Statistics in DLCX Messages (optional)
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Troubleshooting DSP Voice-Quality Statistics in DLCX Messages (optional)
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
4.
end
DETAILED STEPS
Verifying DSP Voice-Quality Statistics in DLCX Messages
Use the following show commands to check your configuration:
SUMMARY STEPS
1.
show call active voice compact
2.
show call active voice brief
3.
show call history voice brief
Step 1
Obtain the call ID by entering the show call active voice compact command in privileged EXEC mode:
Router# show call active voice compactG<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 10.32.1.21:19324Step 2
Check the status of active calls using the call ID obtained from the show call active voice brief command:
Router# show call active voice brief id 11D6<ID>: <start>hs.<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> 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>: 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: 1 SIP call-legs: 0 H323 call-legs: 0 MGCP call-legs: 1 Total call-legs: 2 11D6 : 37530hs.1 +0 pid:0 Originate active dur 00:03:21 tx:1472/29003 rx:1405/27682 Tele 6/4:15 (1): tx:201530/37000/0ms g729r8 noise:-65 acom:90 i/0:-87/-24 dBm 11D6 : 37531hs.1 +-1 pid:0 Originate connecting dur 00:00:00 tx:1403/27642 rx:1472/29003 IP 10.32.1.21:19324 rtt:0ms pl:36000/0ms lost:0/0/0 delay:100/90/110ms g729r8 Telephony call-legs: 1 SIP call-legs: 0 H323 call-legs: 0 MGCP call-legs: 1 Total call-legs: 2Step 3
Verify your configuration with the show call history voice brief command:
Router# show call history voice brief<ID>: <start>hs.<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>MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>last <buf event time>s dur:<Min>/<Max>sFR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>sig:<on/off> <codec> (payload size)ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>sig:<on/off> <codec> (payload size)Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBmTroubleshooting DSP Voice-Quality Statistics in DLCX Messages
Use the debug mgcp packets command and keyword to display statistics reported in the DLCX message generated at the end of the call. The following is sample debug output:
Router# debug mgcp packetsDLCX 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 10.31.1.200:2427 ---> *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=0The following sections provide references related to the DSP Voice-Quality Statistics in DLCX Messages feature.
Related Documents
Related Topic Document TitleHow to configure QoS for Cisco features.
Cisco IOS Release 12.3 mainline roadmap
Cisco IOS Release 12.3 Configuration Guides and Command References
How to configure your Cisco router or access server to support voice, video, and fax applications.
Cisco IOS Voice Configuration Library, Release 12.3
How to use Cisco IOS commands to support voice, video, and fax applications.
Cisco IOS Voice, Video, and Fax Command Reference, Release 12.3
Cisco MGC documentation index
How to configure MGCP
Configuring Media Gateway Control Protocol and Related Protocols
How to configure QoS for voice applications.
How to configure voice ports
Configuring Voice Ports, Release 12.2
Enabling basic management protocols on Cisco access platforms
Cisco IOS Release 12.3
Standards
Standards TitleNo new or modified standards are supported by this feature, and support for existing standards has not been modified by this feature.
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MIBs
RFCs
RFCs TitleNo new or modified RFCs are supported by this feature, and support for existing RFCs has not been modified by this feature.
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Technical Assistance
Command Reference
This section documents new and modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.3 command reference publications.
New Command
Modified Command
debug mgcp
To enable debug traces for Media Gateway Control Protocol (MGCP) errors, events, media, packets, parser, and call admission control (CAC), use the debug mgcp command in privileged EXEC mode. To disable debugging output, use the no form of this command.
debug mgcp [all | errors [endpoint endpoint-name] | events [endpoint endpoint-name] | media [endpoint endpoint-name] | nas | packets [endpoint endpoint-name | input-hex] | parser | src | voipcac]
no debug mgcp [all | errors | events | media | nas | packets | parser | src | voipcac]
Syntax Description
Command Modes
Privileged EXEC
Command History
Usage Guidelines
There is always a performance penalty when using debug commands.
Examples
The following is sample output from the debug mgcp errors, debug mgcp events, debug mgcp media, debug mgcp nas, debug mgcp packets, debug mgcp parser, and debug mgcp src commands and keywords. The debug mgcp all command and keyword would show a compilation of all this output, including the debug mgcp voipcac command and keyword output. Note that using the debug mgcp all command and keyword may severely impact network performance.
The following is sample output from the debug mgcp errors command and keyword:
Router# debug mgcp errorsUnknown network interface typeThe following is sample output from the debug mgcp events command and keyword:
Router# debug mgcp eventsMedia Gateway Control Protocol events debugging is onRouter#1w1d: MGC stat - 10.19.184.65, total=44, succ=7, failed=211w1d: MGCP msg 11w1d: remove_old_under_specified_ack:1w1d: MGC stat - 10.19.184.65, total=44, succ=8, failed=211w1d: updating lport with 2427setup_ipsocket: laddr=172.29.248.193, lport=2427,faddr=10.19.184.65, fport=24271w1d: enqueue_ack: ackqhead=0, ackqtail=0, ackp=1DC1D38, msg=21A037CThe following is sample output from the debug mgcp media command and keyword:
Router# debug mgcp mediaMedia Gateway Control Protocol media events debugging is onRouter#DYNAMIC payload typeDYNAMIC payload type*Jan 1 03:02:13.159:mgcp_verify_supp_reqdet_ev*Jan 1 03:02:13.159:mgcp_verify_supp_signal_ev*Jan 1 03:02:13.159:process_request_ev- callp 635368FC, voice_if 6353C1F8*Jan 1 03:02:13.159:process_detect_ev- callp 635368FC, voice_if 6353C1F8*Jan 1 03:02:13.159:process_signal_ev- callp 635368FC, voice_ifp 6353C1F8*Jan 1 03:02:13.159:mgcp_process_quarantine_mode- callp 635368FC, voice_if 6353C1F8*Jan 1 03:02:13.159:mgcp_process_quarantine_mode- new q mode:process=0, loop=0*Jan 1 03:02:13.179:process_deferred_request_events*Jan 1 03:02:13.479:mgcp_verify_supp_reqdet_ev*Jan 1 03:02:13.479:mgcp_verify_supp_signal_ev*Jan 1 03:02:13.479:process_request_ev- callp 6353BCCC, voice_if 638C3094*Jan 1 03:02:13.479:process_detect_ev- callp 6353BCCC, voice_if 638C3094*Jan 1 03:02:13.479:process_signal_ev- callp 6353BCCC, voice_ifp 638C3094*Jan 1 03:02:13.479:mgcp_process_quarantine_mode- callp 6353BCCC, voice_if 638C3094*Jan 1 03:02:13.479:mgcp_process_quarantine_mode- new q mode:process=0, loop=0*Jan 1 03:02:13.499:process_deferred_request_events*Jan 1 03:02:13.827:mgcp_verify_supp_reqdet_ev*Jan 1 03:02:13.827:mgcp_verify_supp_signal_ev*Jan 1 03:02:13.827:process_request_ev- callp 635368FC, voice_if 6353C1F8*Jan 1 03:02:13.827:process_detect_ev- callp 635368FC, voice_if 6353C1F8*Jan 1 03:02:13.827:process_signal_ev- callp 635368FC, voice_ifp 6353C1F8*Jan 1 03:02:13.827:mgcp_process_quarantine_mode- callp 635368FC, voice_if 6353C1F8*Jan 1 03:02:13.827:mgcp_process_quarantine_mode- new q mode:process=0, loop=0*Jan 1 03:02:13.831:process_deferred_request_events*Jan 1 03:02:23.163:mgcp_cr_and_init_evt_node:$$$ the node pointer 63520B14*Jan 1 03:02:23.163:mgcp_insert_node_to_preprocess_q:$$$enq to preprocess, qhead=63520B14, qtail=63520B14, count 1, evtptr=63520B14*Jan 1 03:02:23.479:mgcp_cr_and_init_evt_node:$$$ the node pointer 63520BA8*Jan 1 03:02:23.479:mgcp_insert_node_to_preprocess_q:$$$enq to preprocess, qhead=63520BA8, qtail=63520BA8, count 1, evtptr=63520BA8The following is sample output for the debug mgcp nas command and keyword, with the debug mgcp packets command and keyword enabled as well:
Router# debug mgcp nasMedia Gateway Control Protocol nas pkg events debugging is onRouter# debug mgcp packetsMedia Gateway Control Protocol packets debugging is onRouter#01:49:14:MGCP Packet received -CRCX 58 S7/DS1-0/23 MGCP 1.0X:57M:nas/dataC:3L:b:64, nas/bt:modem, nas/cdn:3000, nas/cgn:1000mgcp_parse_conn_mode :string past nas = datamgcp_chq_nas_pkg:Full string:nas/bt:modemmgcp_chq_nas_pkg:string past slash:btmgcp_chq_nas_pkg:string past colon:modemmgcp_chq_nas_pkg:Full string:nas/cdn:3000mgcp_chq_nas_pkg:string past slash:cdnmgcp_chq_nas_pkg:string past colon:3000mgcp_chq_nas_pkg:Full string:nas/cgn:1000c5400#mgcp_chq_nas_pkg:string past slash:cgnmgcp_chq_nas_pkg:string past colon:1000CHECK DATA CALL for S7/DS1-0/23mgcpapp_xcsp_get_chan_cb -Found - Channel state IdleCRCX Recvmgcpapp_endpt_is_data:endpt S7/DS1-0/23, slot 7, port 0 chan 23mgcpapp_data_call_hnd:mgcpapp_xcsp_get_chan_cb -Found - Channel state Idlebw=64, bearer=E1,cdn=3000,cgn=1000The following is sample output from the debug mgcp packets command and keyword:
Router# debug mgcp packetsMedia Gateway Control Protocol packets debugging is onRouter#1w1d: MGCP Packet received -DLCX 408631346 * MGCP 0.11w1d: send_mgcp_msg, MGCP Packet sent --->1w1d: 250 408631346<---The following is sample output from the debug mgcp parser command and keyword:
Router# debug mgcp parserMedia Gateway Control Protocol parser debugging is onRouter#1w1d: -- mgcp_parse_packet() - call mgcp_parse_header- mgcp_parse_header()- Request Verb FOUND DLCX- mgcp_parse_packet() - out mgcp_parse_header- SUCCESS: mgcp_parse_packet()- MGCP Header parsing was OK- mgcp_val_mandatory_parms()- SUCCESS: mgcp_parse_packet()- END of Parsing1w1d: -- mgcp_build_packet()-1w1d: - mgcp_estimate_msg_buf_length() - 87 bytes needed for header- mgcp_estimate_msg_buf_length() - 87 bytes needed after checking parameter lines- mgcp_estimate_msg_buf_length() - 87 bytes needed after checking SDP lines- SUCCESS: MGCP message building OK- SUCCESS: END of buildingThe following is sample output from the debug mgcp src command and keyword:
Router# debug mgcp srcMedia Gateway Control Protocol System Resource Check CAC debugging is onRouter#00:14:08: setup_indication: Set incoming_call flag=TRUE in voice_if00:14:08: send_mgcp_msg, MGCP Packet sent --->00:14:08: NTFY 11 aaln/S1/1@Router MGCP 0.1N: emu@[1.4.173.1]:51665X: 35O: hd<---00:14:08: MGCP Packet received -200 11 hello00:14:08: MGCP Packet received -RQNT 42 aaln/S1/1 MGCP 0.1N: emu@[10.40.155.1]:51665X: 41R: D/[0-9*#T](d), huS: dlD: (911|xxxx)00:14:08: send_mgcp_msg, MGCP Packet sent --->00:14:08: 200 42 OK<---00:14:12: send_mgcp_msg, MGCP Packet sent --->00:14:12: NTFY 12 aaln/S1/1@Router MGCP 0.1N: emu@[10.40.155.1]:51665X: 41O: D/2222<---00:14:12: MGCP Packet received -200 12 phone-number ok00:14:12: MGCP Packet received -CRCX 44 aaln/S1/1 MGCP 0.1N: emu@[10.40.155.1]:51665C: 3X: 43R: hu(n)M: recvonlyL: a:G.711u,p:5,e:off,s:off00:14:12: mgcp_setup_conn_check_system_resource: System resource check successful00:14:12: mgcp_voice_crcx: System resource is available00:14:12: mgcp_set_call_counter_control: Incoming call with 1 network leg, flag=FALSE00:14:12: send_mgcp_msg, MGCP Packet sent --->00:14:12: 200 44I: 4v=0o=- 4 0 IN IP4 10.20.120.1s=Cisco SDP 0c=IN IP4 10.20.120.1t=0 0m=audio 16404 RTP/AVP 0<---00:14:13: MGCP Packet received -MDCX 48 aaln/S1/1 MGCP 0.1N: emu@[10.40.155.1]:51665C: 3I: 4X: 47M: recvonlyR: huL: a:G.711u,p:5,e:off,s:offv=0o=- 4 0 IN IP4 1.4.120.3s=Cisco SDP 0c=IN IP4 10.4.110.3t=0 0m=audio 16384 RTP/AVP 000:14:13: mgcp_modify_conn_check_system_resource: System resource check successful00:14:13: mgcp_modify_connection: System resource is available00:14:13: send_mgcp_msg, MGCP Packet sent --->00:14:13: 200 48 OK<---00:14:20: MGCP Packet received -MDCX 52 aaln/S1/1 MGCP 0.1N: emu@[10.40.155.1]:51665C: 3I: 4X: 51M: sendrecvR: huL: a:G.711u,p:5,e:off,s:off00:14:20: mgcp_modify_conn_check_system_resource: System resource check successful00:14:20: mgcp_modify_connection: System resource is available00:14:20: send_mgcp_msg, MGCP Packet sent --->00:14:20: 200 52 OK<---00:14:34: MGCP Packet received -DLCX 56 aaln/S1/1 MGCP 0.1X: 55N: emu@[10.40.155.1]:51665C: 3I: 4R: hu00:14:34: send_mgcp_msg, MGCP Packet sent --->00:14:34: 250 56P: PS=1382, OS=110180, PR=1378, OR=109936, PL=63484, JI=520, LA=2<---00:14:36: mgcp_reset_call_direction: Reseting incoming_call flag=FALSE in voice_if00:14:36: send_mgcp_msg, MGCP Packet sent --->00:14:36: NTFY 13 aaln/S1/1@tlkrgw1 MGCP 0.1N: emu@[10.40.155.1]:51665X: 55O: hu<---The following example displays statistics reported in the DLCX message generated at the end of a call:
Router# debug mgcp packetsDLCX 311216 s6/ds1-4/1@as5400a MGCP 0.1 C: 48A4B I: 2 R: S: X: 4BFAF *May 5 10:38:51.643: send_mgcp_msg, MGCP Packet sent to 10.31.1.200:2427 ---> *May 5 10:38:51.643: 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=0Table 2 describes the significant fields shown in the display.
mgcp voice-quality-stats
To enable voice-quality statistics reporting for the Media Gateway Control Protocol (MGCP), use the mgcp voice-quality-stats command in global configuration mode. To turn off voice-quality statistics reporting, use the no form of this command.
mgcp voice-quality-stats [priority<value>] | [all]
no mgcp voice-quality-stats [priority<value>] | [all]
Syntax Description
priority<value>
Selects numeric parameters 1 or 2 to indicate priority.
all
Selects all VQ parameters.
Command Default
Voice-quality statistics reporting is turned off.
Command Modes
Global configuration
Command History
Release Modification12.3(3)
This command was introduced.
12.4(4)T
The priority and all keywords were introduced.
Usage Guidelines
•
The request for digital signal processor (DSP) statistics is controlled by the RTP Control Protocol (RTCP) statistics polling interval. The polling interval is configurable by entering the ip rtcp report interval command. Statistics are polled every 5 seconds by default.
Note
The Cisco PGW 2200 must have a patch the supports DSP statistics in order to collect data in the call detail records (CDRs).
•
This command does not generate any output on the console; it adds additional quality statistics parameters in the MGCP Delete Connection (DLCX) ACK message that is sent to the call agent.
Cisco IOS Release 12.4(4)T supports only priority levels 1 and 2.
•
The keyword priority uses a value of 1 or 2 to indicate the priority of the parameters.
Note
Choosing priority 2 is similar to using the keyword all where all the paramters are selected.
The corresponding set of VQ parameters are sent in the MGCP DLCX message based on the priority selected.
Examples
The following example enables voice-quality statistics reporting for MGCP:
Router> enableRouter# configure terminalRouter(config)# mgcp voice-quality-statsRouter(config)# endThe following example shows the VQ parameters selected for priority 1:
mgcp voice-quality-stats priority 116:38:20.461771 10.0.5.130:2427 10.0.5.133:2427 MGCP...... -> 250 1133 OKP: PS=0, OS=0, PR=0, OR=0, PL=0, JI=65, LA=0DSP/TX: PK=118, SG=0, NS=1, DU=28860, VO=2350DSP/RX: PK=0, SG=0, CF=0, RX=28860, VO=0, BS=0, LP=0, BP=0DSP/PD: CU=65, MI=65, MA=65, CO=0, IJ=0DSP/LE: TP=0, RP=0, TM=0, RM=0, BN=0, ER=0, AC=0DSP/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=0The following example shows all the VQ parameters selected for the keyword all:
mgcp voice-quality-stats all16:38:20.461771 10.0.5.130:2427 10.0.5.133:2427 MGCP...... -> 250 1133 OKP: PS=0, OS=0, PR=0, OR=0, PL=0, JI=65, LA=0DSP/TX: PK=118, SG=0, NS=1, DU=28860, VO=2350DSP/RX: PK=0, SG=0, CF=0, RX=28860, VO=0, BS=0, LP=0, BP=0DSP/PD: CU=65, MI=65, MA=65, CO=0, IJ=0DSP/PE: PC=0, IC=0, SC=0, RM=0, BO=0, EE=0DSP/LE: TP=0, RP=0, TM=0, RM=0, BN=0, ER=0, AC=0DSP/ER: RD=0, TD=0, RC=0, TC=0DSP/IC: IC=0DSP/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=0Related Commands
Command Descriptiondebug mgcp
Enables debug traces for MGCP errors, events, media, packets, parser, and CAC.
ip rtcp report interval
Configures the RTCP statistics polling interval.
Glossary
AAL2—ATM adaptation layer 2.
ASP—application service provider.
CA—call agent.
CAC—call admission control.
CAS—channel-associated signaling.
CDR—call detail record.
CLI—command-line-interface.
DCLX—MGCP Delete Connection message.
DSP—digital signal processor.
FTP—File Transfer Protocol.
NAS—network access server.
PVC—permanent virtual circuit.
RSVP—Resource Reservation Protocol.
RTCP—RTP Control Protocol. Protocol that monitors the QoS of an IPv6 RTP connection and conveys information about the ongoing session.
RTP—Real-Time Transport Protocol.
SRC—system resource check.
TDM—time-division multiplexing.
VPN—virtual private network.
WFQ—weighted fair queueing.
Note
Refer to Internetworking Terms and Acronyms for terms not included in this glossary.
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