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Cisco IOS Software Releases 12.3 Mainline

DSP Voice Quality Statistics in DLCX Messages

Table Of Contents

DSP Voice-Quality Statistics in DLCX Messages

Contents

Prerequisites for DSP Voice-Quality Statistics in DLCX Messages

Information About DSP Voice-Quality Statistics in DLCX Messages

Cisco PGW 2200

MGCP

Voice Quality Statistics

Quality of Service for Voice

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

Related Documents

Standards

MIBs

RFCs

Technical Assistance

Command Reference

debug mgcp

mgcp voice-quality-stats

Glossary


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
Modification

12.3(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

Prerequisites for DSP Voice-Quality Statistics in DLCX Messages

Information About DSP Voice-Quality Statistics in DLCX Messages

How to Configure DSP Voice-Quality Statistics in DLCX Messages

Command Reference

Glossary

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:

Cisco PGW 2200

MGCP

Voice Quality Statistics

Quality of Service for Voice

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.


Caution There 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:

International and national transit networks

Dial access

Application service provider (ASP) termination

Managed business voice applications

Managed voice virtual private networks (VPNs)

PSTN access for hosted and managed IP telephony

Residential voice applications

PSTN access for voice over broadband networks

Network clearinghouse applications

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.

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

DSP Technology
Platform
Voice Quality Statistics

MSA V6

Cisco AS5350, Cisco AS5350XM, Cisco AS5400, Cisco AS5400HPX, Cisco 5400XM, and Cisco AS5850 with a 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 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:

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.

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.

Customized services—QoS enables Internet service providers (ISPs) to offer carefully customized grades of service differentiation to their customers.

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.

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:

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

Cisco IOS Release 12.4(4)T introduces these new voice quality parameters:

DSP/EC : Endpoint Configuration

DSP/KF : MOS/K-Factor Statistics

DSP/CS: Concealment Statistics

DSP/RF—R-Factor Statistics

DSP/UC: User Concealment Statistics

DSP/DL: Delay 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.

CI—Codec ID

A string or number that identifies the voice codec which is currently used in the call.

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.

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.

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.

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.

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.

JD—Jitter buffer mode

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 : 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.

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.

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 eight second interval.

BS—Baseline (maximum) k-factor score

K-factor score that can be obtained for the provisioned codec.

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.

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.

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: Concealment Statistics

It measures packet (frame) loss and its effect on voice quality in an impaired network.

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.

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 seconds

The duration of time during which some concealment is observed.

CT—Total concealment time in milliseconds

The total duration of time during which concealment is observed after starting a call.

TT—Total speech time in milliseconds

The duration of time during which active speech is observed after starting a call.

OK—Ok seconds

The duration of time in seconds during which no concealment is observed.

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.

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

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

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: User Concealment Statistics

U1—User concealment seconds 1 count (UCS1)

U2—User concealment seconds 2 count (UCS2)

T1—UCS1 threshold in ms

T2—UCS2 threshold in ms

DSP/DL: Delay Statistics

RT—Round trip delay

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.

Configuring DSP Voice-Quality Statistics in DLCX Messages (required)

Verifying DSP Voice-Quality Statistics in DLCX Messages (optional)

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

 
Command or Action
Purpose

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[priority<1|2>]|[all]

Example:

Router(config)# mgcp voice-quality-stats 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 where the output shows 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, DSP/DL.

Step 4 

end

Example:

Router(config)# end

Completes the configuration.

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 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 10.32.1.21:19324

Step 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: 2

Step 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>s
 FR <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>dBm

Troubleshooting 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 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 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=0

The following sections provide references related to the DSP Voice-Quality Statistics in DLCX Messages feature.

Related Documents

Related Topic
Document Title

How to configure QoS for Cisco features.

Cisco IOS Quality of Service Solutions Configuration Guide

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

Cisco Media Gateway Controllers

How to configure MGCP

Configuring Media Gateway Control 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, Release 12.2

Enabling basic management protocols on Cisco access platforms

Enabling Management Protocols: NTP, SNMP, and Syslog

Cisco IOS Release 12.3

Release notes index, Cisco IOS Release 12.3


Standards

Standards
Title

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


MIBs

MIBs
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 IOS releases, and feature sets, use Cisco MIB Locator found at the following URL:

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


RFCs

RFCs
Title

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


Technical Assistance

Description
Link

Technical Assistance Center (TAC) home page, containing 30,000 pages of searchable technical content, including links to products, technologies, solutions, technical tips, and tools. Registered Cisco.com users can log in from this page to access even more content.

http://www.cisco.com/public/support/tac/home.shtml


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

mgcp voice-quality-stats

Modified Command

debug mgcp

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

all

(Optional) Enables debugging traces for MGCP errors, events, media, packets, parser and builder, and CAC.

errors

(Optional) Enables debugging traces for MGCP errors.

endpoint endpoint-name

(Optional) Enables debugging traces for MGCP errors, events, media, or packets per endpoint.

events

(Optional) Enables debugging traces for MGCP events.

media

(Optional) Enables debugging traces for MGCP tone and signal events.

nas

(Optional) Enables debugging traces for MGCP network access server (NAS) (data) events.

packets

(Optional) Enables debugging traces for MGCP packets.

input-hex

(Optional) Enables debugging traces for MGCP input packets in hexadecimal values.

parser

(Optional) Enables debugging traces for MGCP parser and builder.

src

(Optional) Enables debugging traces for MGCP system resource check (SRC) CAC information.

voipcac

(Optional) Enables debugging traces for the VoIP CAC process at the MGCP application layer.


Command Modes

Privileged EXEC

Command History

Release
Modification

12.1(1)T

This command was introduced.

12.1(3)T

Additional information was displayed for the gateways.

12.1(5)XM

The output was modified to display parameters for the MGCP channel-associated signaling (CAS) PBX and ATM adaptation layer 2 (AAL2) permanent virtual circuit (PVC) features.

12.2(2)XA

The media keyword was added. The endpoint endpoint-name keyword and argument were added as options for the errors, events, media, and packets keywords. The input-hex keyword option was added for the packets keyword.

12.2(2)T

The output was further modified to display parameters for the MGCP CAS PBX and AAL2 PVC features.

12.2(2)XB

The nas keyword and the src and voipcac keywords were added. (Refer to MGCP VoIP Call Admission Control in Cisco IOS Release 12.2(2)XB.)

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T.

Note The nas keyword was not integrated into Cisco IOS Release 12.2(8)T.

12.2(11)T

The command was implemented on the Cisco AS5350, Cisco AS5400, and Cisco AS5850.

12.2(13)T

Support for this command was implemented in Cisco 7200 series.

12.3(3)

DSP statistics were reported in the MGCP Delete Connection (DLCX) message.


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 errors 

Unknown network interface type 

The following is sample output from the debug mgcp events command and keyword:

Router# debug mgcp events 

Media Gateway Control Protocol events debugging is on 
Router# 
1w1d: MGC stat - 10.19.184.65, total=44, succ=7, failed=21 
1w1d: MGCP msg 1 
1w1d: remove_old_under_specified_ack: 
1w1d: MGC stat - 10.19.184.65, total=44, succ=8, failed=21 
1w1d: updating lport with 2427setup_ipsocket: laddr=172.29.248.193, lport=2427, 
faddr=10.19.184.65, fport=2427 
1w1d: enqueue_ack: ackqhead=0, ackqtail=0, ackp=1DC1D38, msg=21A037C 

The following is sample output from the debug mgcp media command and keyword:

Router# debug mgcp media 

Media Gateway Control Protocol media events debugging is on 
Router# 
DYNAMIC payload type 
DYNAMIC 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=63520BA8 

The 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 nas 

Media Gateway Control Protocol nas pkg events debugging is on 
Router# debug mgcp packets 
Media Gateway Control Protocol packets debugging is on 
Router# 
01:49:14:MGCP Packet received - 
CRCX 58 S7/DS1-0/23 MGCP 1.0 
X:57 
M:nas/data 
C:3 
L:b:64, nas/bt:modem, nas/cdn:3000, nas/cgn:1000 
mgcp_parse_conn_mode :string past nas = data 
mgcp_chq_nas_pkg:Full string:nas/bt:modem 
mgcp_chq_nas_pkg:string past slash:bt 
mgcp_chq_nas_pkg:string past colon:modem 
mgcp_chq_nas_pkg:Full string:nas/cdn:3000 
mgcp_chq_nas_pkg:string past slash:cdn 
mgcp_chq_nas_pkg:string past colon:3000 
mgcp_chq_nas_pkg:Full string:nas/cgn:1000 
c5400# 
mgcp_chq_nas_pkg:string past slash:cgn 
mgcp_chq_nas_pkg:string past colon:1000 
CHECK DATA CALL for S7/DS1-0/23 
mgcpapp_xcsp_get_chan_cb -Found - Channel state Idle 
CRCX Recv 
mgcpapp_endpt_is_data:endpt S7/DS1-0/23, slot 7, port 0 chan 23 
mgcpapp_data_call_hnd:mgcpapp_xcsp_get_chan_cb -Found - Channel state Idle 
bw=64, bearer=E1,cdn=3000,cgn=1000 

The following is sample output from the debug mgcp packets command and keyword:

Router# debug mgcp packets 

Media Gateway Control Protocol packets debugging is on 
Router# 
1w1d: MGCP Packet received - 
DLCX 408631346 * MGCP 0.1 
1w1d: 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 parser 

Media Gateway Control Protocol parser debugging is on 
Router# 
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 Parsing 
1w1d: -- 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 building 

The following is sample output from the debug mgcp src command and keyword:

Router# debug mgcp src 

Media Gateway Control Protocol System Resource Check CAC debugging is on 
Router# 
00:14:08: setup_indication: Set incoming_call flag=TRUE in voice_if 
00:14:08: send_mgcp_msg, MGCP Packet sent ---> 
00:14:08: NTFY 11 aaln/S1/1@Router MGCP 0.1 
N: emu@[1.4.173.1]:51665 
X: 35 
O: hd 
<--- 
00:14:08: MGCP Packet received - 
200 11 hello 
00:14:08: MGCP Packet received - 
RQNT 42 aaln/S1/1 MGCP 0.1 
N: emu@[10.40.155.1]:51665 
X: 41 
R: D/[0-9*#T](d), hu 
S: dl 
D: (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 0.1 
N: emu@[10.40.155.1]:51665 
X: 41 
O: D/2222 
<--- 
00:14:12: MGCP Packet received - 
200 12 phone-number ok 
00:14:12: MGCP Packet received - 
CRCX 44 aaln/S1/1 MGCP 0.1 
N: emu@[10.40.155.1]:51665 
C: 3 
X: 43 
R: hu(n) 
M: recvonly 
L: a:G.711u,p:5,e:off,s:off 
00:14:12: mgcp_setup_conn_check_system_resource: System resource check successful 
00:14:12: mgcp_voice_crcx: System resource is available 
00:14:12: mgcp_set_call_counter_control: Incoming call with 1 network leg, flag=FALSE 
00:14:12: send_mgcp_msg, MGCP Packet sent ---> 
00:14:12: 200 44 
I: 4 
v=0 
o=- 4 0 IN IP4 10.20.120.1 
s=Cisco SDP 0 
c=IN IP4 10.20.120.1 
t=0 0 
m=audio 16404 RTP/AVP 0 
<--- 
00:14:13: MGCP Packet received - 
MDCX 48 aaln/S1/1 MGCP 0.1 
N: emu@[10.40.155.1]:51665 
C: 3 
I: 4 
X: 47 
M: recvonly 
R: hu 
L: a:G.711u,p:5,e:off,s:off 
v=0 
o=- 4 0 IN IP4 1.4.120.3 
s=Cisco SDP 0 
c=IN IP4 10.4.110.3 
t=0 0 
m=audio 16384 RTP/AVP 0 
00:14:13: mgcp_modify_conn_check_system_resource: System resource check successful 
00:14:13: mgcp_modify_connection: System resource is available 
00:14:13: send_mgcp_msg, MGCP Packet sent ---> 
00:14:13: 200 48 OK 
<--- 
00:14:20: MGCP Packet received - 
MDCX 52 aaln/S1/1 MGCP 0.1 
N: emu@[10.40.155.1]:51665 
C: 3 
I: 4 
X: 51 
M: sendrecv 
R: hu 
L: a:G.711u,p:5,e:off,s:off 
00:14:20: mgcp_modify_conn_check_system_resource: System resource check successful 
00:14:20: mgcp_modify_connection: System resource is available 
00:14:20: send_mgcp_msg, MGCP Packet sent ---> 
00:14:20: 200 52 OK 
<--- 
00:14:34: MGCP Packet received - 
DLCX 56 aaln/S1/1 MGCP 0.1 
X: 55 
N: emu@[10.40.155.1]:51665 
C: 3 
I: 4 
R: hu 
00:14:34: send_mgcp_msg, MGCP Packet sent ---> 
00:14:34: 250 56 
P: 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_if 
00:14:36: send_mgcp_msg, MGCP Packet sent ---> 
00:14:36: NTFY 13 aaln/S1/1@tlkrgw1 MGCP 0.1 
N: emu@[10.40.155.1]:51665 
X: 55 
O: hu 
<--- 

The following example displays statistics reported in the DLCX message generated at the end of a call:

Router# debug mgcp packets

DLCX 311216 s6/ds1-4/1@as5400a MGCP 0.1 
C: 48A4B 
I: 2 
R:  
S:  
X: 4BFAF 
*May 5 10:38:51.643: 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=0

Table 2 describes the significant fields shown in the display.

Table 2 debug mgcp packets Field Descriptions 

Field
Description

DSP/TX

DSP transmission (TX) statistics:

PK—Transmission (TX) packets

SG—Signaling packets

NS—Noise packets

DU—Transmission duration

VO—Voice transmission duration

DSP/RX

DSP receive (RX) statistics:

PK—Voice packets

SG—Signaling packets

CF—Comfort noise packets

RX—Receive duration

VO—Voice receive duration

BS—Bad sequence

BP—Bad protocol

LP—Late packets

DSP/PD

DSP playout delay (PD) statistics:

CU—Current

MI—Minimum

MA—Maximum

CO—clock_offset

IJ—Interarrival jitter

DSP/PE

DSP playout error (PE) statistics:

PC—Predictive concealment

IC—Interpolative concealment

SC—Silence concealment

RM—Retroactive memory update

BO—Buffer overflow

EE—Number of talkspurt endpoint errors

DSP/LE

DSP level (LE) statistics:

TP—Transmission power in 0.1 dBm units

TX—Transmission mean in 0.1 linear PCM units

RP—Receive power in 0.1 dBm units

RM—Receive mean in 0.1 linear PCM units

BN—Background noise

ER—Echo return loss (ERL) level

AC—Acom level

TA—Current transmit activity

RA—Current receive activity

DSP/ER

DSP error statistics:

RD—Receive dropped

TD—Transmission dropped

RC—Receive control

TC—Transmission control

DSP/IC

ICPIF value for measuring voice-quality.


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
Modification

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


The following example shows the VQ parameters selected for priority 1:

mgcp voice-quality-stats priority 1

16:38:20.461771 10.0.5.130:2427 10.0.5.133:2427 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 all the VQ parameters selected for the keyword all:

mgcp voice-quality-stats all 

16:38:20.461771 10.0.5.130:2427 10.0.5.133:2427 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

Related Commands

Command
Description

debug 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.