Monitor Voice Quality

Overview

Call quality statistics in CUBE, such as packet loss, jitter, and round trip delay can be added to the call detail record (CDR), and these voice metrics can be calculated in IOS. For more information, refer to Voice Quality Enhancements on Cisco Unified Border Element.

The call quality statistics feature is enhanced to provide the following capabilities:

  • Enable or disable Quality of Service (QoS) for CUBE.

  • Enable or disable Real-time Transport Protocol (RTP) Control Protocol (RTCP) passthrough.

  • Configure call quality criteria parameters.

Call quality configuration parameters include max_dropout, max_reorder, and clock_rate. A maximum of three codecs (codec_number, payload_type, clock_rate) per media flow is collected by the PI and sent to CPP, which uses these values in statistics calculation. Calculated statistics such as Jitter, Packet Loss, and Delay are then fetched from the CPP to the CDR. These statistics can be viewed in the command line interface.

The CDR has the following data per call leg of the call:

  • Packet Loss-Calculated based on methods shown in RFC3550. The RTCP sender/receiver reports are recalculated, and not just copied from the inbound leg to the outbound leg.

  • Delay-Calculated based on timestamp received or timestamp of packets sent.

  • Jitter-Variation of delay.

For more information on how to calculate the voice quality metrics related to media(voice) quality, such as conversational mean opinion score (MOS), jitter, and so on, see http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-call-monitoring.html.

The VQM (Voice Quality Monitor) gives information on the voice quality metrics. The VQM on Cisco IOS XE platforms enables statistics gathering based on the received RTCP packets. From these statistics, a voice quality measurement is developed to show the quality of the call. The output is in a simple format, using a system of good, poor, and bad types of ratings.

The following metrics exists in Call Detail Record (CDR) and Management Information Base (MIB) in CUBE, indicating voice quality:

  1. MOSQe (conversational quality MOS)

  2. Round-trip-delay.

  3. Receive-delay (current jitter buffer size).

  4. Packet-Loss-Rate.

The CDR is sent at the end of a call if AAA accounting is configured.

A CDR example is as follows:

<MOS-Con>4.4072</MOS-Con>

<round-trip-delay>1 ms</round-trip-delay>

<receive-delay>64 ms</receive-delay>

<voice-quality-total-packet-loss>0.0000 %</ voice-quality-total-packet-loss>

VQM Metrics

The following are the metrics exported by VQM:

Table 1. Quality Metrics

IOS VQM, Voice/Audio Description Quality Metric

Description

MOS-Con

The conversational quality MOS. Conversational quality indicates the impact of the quality of the transmission on the dynamics of conversational exchanges between two parties; such metrics take into account delay, echo, and recency.

round-trip-delay

The instantaneous round-trip delay. This may be obtained from the RTCP SR reports.

receive-delay

The minimum delay that will be applied to the packets received when using an adaptive jitter buffer.

voice-quality-total-packet-loss

The total number of packets lost by the jitter buffer in the RTP stream.

Feature Information

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 Voice Quality Monitoring and Voice Quality Statistics

Feature Name

Releases

Feature Information

Voice Quality Statistics

Cisco IOS XE Everest 16.5.1b

Voice quality statistics provides information about the quality of the voice TDM-IP call. This feature is already implemented on ISR-G2, and the feature gap is filled in ISR 4000 series.

Voice Quality Monitoring

Cisco IOS XE Denali 16.3.1

The Voice Quality Monitoring (VQM) feature provides information on the voice quality metrics related to media (voice) quality, such as conversational mean opinion score (MOS), packet loss rate, and so on. VQM enables you to monitor the quality of calls traversing your VoIP network, and you can diagnose the cause of voice quality issues and troubleshoot them.

Prerequisites

The following commands must be executed to configure the voice quality metrics:

  • callmonitor

  • rtcp all-pass-through

  • media statistics

  • media bulk-stats

  • call-quality

    • max-dropout 2

    • max-reorder 2

Restrictions

  • Only SIP-to-SIP call quality statistics calculation is supported.

  • The RTCP field is not recalculated, as it is end-to-end statistics.

  • The round trip delay is only retrieved by RTCP, which means the round trip delay is not calculated if there is no related RTCP.

  • Only three codec types are supported for one media flow to calculate the jitter; considering the data path performance, these three codecs would be the maximum number in one cache line.

  • Only one RTP synchronization source (SSRC) is supported concurrently per media flow, which is indicated in the m-line of the session description protocol (SDP).

  • Round trip delay calculation for transcoding calls is not supported.

  • VQM MOS values are not calculated for DSP based calls.

  • MOS value shows 0 if endpoint does not send RTCP packets.

  • The voice quality statistics covers only the TDM-IP call. The implementation focuses on filling the following statistics based on query response from DSP for TDM-SIP and TDM-H323 call:

    • RoundTripDelay

    • GapFillWithSilence

    • GapFillWithPrediction

    • GapFillWithInterpolation

    • GapFillWithRedundancy

    • HiWaterPlayoutDelay

    • LoWaterPlayoutDelay

    • PlayDelayJitter

Configure Voice Quality Monitoring

Enable Media Statistics Globally

Perform this task to globally enable media statistics in voice-service configuration mode to estimate the values for packet loss, jitter, and round-trip time.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice service voip
  4. media statistics
  5. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:


 Device> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Device# configure terminal

Enters global configuration mode.

Step 3

voice service voip

Example:

Device(config)# voice service voip

Enters voice service VoIP configuration mode.

Step 4

media statistics

Example:

Device(conf-voi-serv)# media statistics

Enables media statistics to estimate the values of packet loss, jitter, and Round Trip Time (RTT) statistics.

  • The statistics are displayed using the show voice history and show call active voice commands.

  • If the media statistics command is disabled, the values will be zero.

Step 5

end

Example:

Device(conf-voi-serv)# end

Returns to privileged EXEC mode.

Example: Configuring Media Statistics Globally


Device> enable
Device# configure terminal
Device(config)# voice service voip
Device(conf-voi-serv)# media statistics
Device(conf-voi-serv)# end

Example: CDR Enabled MOS Output

At the end of a call, the MOSQe output is displayed in CDR only if the debug radius accounting is enabled.

The show log | sec MOS-Con command displays the MOS-Con value as shown below: 

Device# show log | sec MOS-Con 

*Jan 21 22:31:42.313: RADIUS:   Cisco AVpair       [1]   16  "MOS-Con=4.2312"
*Jan 21 22:31:42.313: RADIUS:   Cisco AVpair       [1]   16  "MOS-Con=4.2312"

Report End-of-Call Statistics

The reporting End-of-Call statistics in Session Initiation Protocol (SIP) BYE message feature enables you to send call statistics to a remote end when a call terminates. The call statistics are sent as a new header in the BYE message or in the 200 OK message (response to BYE message). The statistics include Real-time Transport Protocol (RTP) packets sent or received, total bytes sent or received, total number of packets that are lost, delay jitter, round-trip delay, and call duration.

This feature enables Cisco Unified Border Element (Cisco UBE) to use the call statistics to update the call data records in Cisco Unified Communications Manager (Cisco UCM) or Cisco Unified Communications Manager Express (Cisco UCME).

Use the media bulk-stats and media stats-disconnect commands in voice service voip configuration mode to enable reporting of RTP statistics during call disconnect in SIP BYE message feature on Cisco Unified Border Element.


Note

To disable reporting End-of-Call statistics in SIP BYE message at the global level, see Disabling Reporting End-of-Call Statistics in SIP BYE Message.

Enable End-of-Call Statistics Reporting in SIP BYE Message at the Global Level

Perform this task to enable the support for reporting End-of-Call statistics in SIP BYE message feature at the global level.

The support for reporting End-of-Call statistics in SIP BYE message feature is enabled by default on Cisco UBE. Hence, to disable the feature, you must modify the SIP profiles to remove the P-RTP-Stat SIP header from the request and the response messages and then configure the modified SIP profile on the Cisco UBE.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice service voip
  4. media bulk-stats
  5. media stats-disconnect

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router>enable

Enables privileged EXEC mode.

Note

 

Enter your password if prompted.

Step 2

configure terminal

Example:

Router#configure terminal
Router(config)#

Enters global configuration mode.

Step 3

voice service voip

Example:

Router(config)# voice servive voip
Router(conf-voi-serv)#

Specifies VoIP as the voice encapsulation method and enters voice-service configuration mode.

Step 4

media bulk-stats

Example:

Router(conf-voi-serv)#media bulk-stats
Router(conf-voi-serv)#

Enables a periodic process to retrieve bulk call statistics.

Step 5

media stats-disconnect

Example:

Router(conf-voi-serv)#media stats-disconnect
Router(conf-voi-serv)#

Enables RTP statistics reporting during call disconnect.
Example: Enable End-of-Call Statistics Reporting

Router>enable
Router#configure terminal
Router(config)# voice servive voip
Router(conf-voi-serv)#media bulk-stats
Router(conf-voi-serv)#media stats-disconnect
Router(conf-voi-serv)#

Verify

Perform this task to verify the configuration for voice quality monitoring. The show commands can be entered in any order.

SUMMARY STEPS

  1. enable
  2. show call active voice | include LostPackets
  3. show call active voice | include ReceiveDelay
  4. show call active voice brief | sec RTT
  5. show call active voice stats | sec MC

DETAILED STEPS

Step 1

enable

Enables privileged EXEC mode.

Example:

Device> enable

Step 2

show call active voice | include LostPackets

Displays statistics on the CUBE if the Voice Quality Metrics feature is configured.

Example:

Device# show call active voice |  include LostPackets 

LostPackets=0
LostPackets=0

Step 3

show call active voice | include ReceiveDelay

Displays statistics on the CUBE if the Voice Quality Metrics feature is configured.

Example:

Device# show call active voice |  include ReceiveDelay 

ReceiveDelay=0
ReceiveDelay=0

Step 4

show call active voice brief | sec RTT

Displays a truncated version of call information for voice calls in CUBE if the Voice Quality Metrics feature is configured.

Note

 

This command is not applicable for TDM-IP voice calls.

Example:

Device# show call active voice brief |  sec RTT 

 IP 173.39.65.81:7078     SRTP: off  rtt:12ms pl:0/0ms  lost:0/0/0  delay:0/0/0ms   g711ulaw   TextRelay: off   Transcoded: No   ICE: Off
 IP 10.127.17.141:18920   SRTP: off  rtt:12ms pl:0/0ms  lost:0/0/0  delay:0/0/0ms   g711ulaw   TextRelay: off   Transcoded: No   ICE: Off

Step 5

show call active voice stats | sec MC

Displays R-Factor Statistics (G.107 MOS) on the CUBE if the Voice Quality Metrics feature is configured. A sample output is provided below for a voice call using G.711ulaw, VAD on, and at 5 percent packet loss rate.

Example:

Device# show call active voice stats |  sec MC 

DSP/RF: ML=, MC=, R1=, R2=, IF=, ID=, IE=, BL=, R0=, VR=
DSP/RF: ML=4.2346, MC=4.2346, R1=92, R2=92, IF=0, ID=0, IE=0, BL=0, R0=93, VR=2.0
DSP/RF: ML=4.2346, MC=4.2346, R1=92, R2=92, IF=0, ID=0, IE=0, BL=0, R0=93, VR=2.0

The following is an example output for the SNMP MIB:
cmqVoIPCallActiveRxPred107RMosConv.8520964.1 = 423 (MC)

For more information on the SNMP MIB "cmqVoIPCallActiveRxPred107RMosConv", see SNMP Object Navigator.

In the sample output, the following can be noted:

  • ML for codec G.711ulaw is 4.2346.

  • MC for codec G.711ulaw is 4.2346.

  • IE for codec G.711 is 0.

  • R0 is 93.

The following table defines the abbreviations used in the sample output.

Table 3. Router Output Definitions for the show call active voice stats command

Type

Abbreviation Definition
DSP/RF: R-Factor Statistics (G.107 MOS) ML R-factor MOS listening quality objective
MC R-factor MOS-CQE
R1 R-factor for LQ profile1
R2 R-factor for LQ
IF Effective codec impairment (IeEff )
ID Delay factors
IE Codec baseline score (Ie)
BL Codec baseline (Bpl)
R0 Nominal value for R0 (default)
VR R-Factor version

Tips to Troubleshoot

Use the following debug commands to troubleshoot the Voice Quality Monitoring feature:

  • debug voip rtp packets

  • debug performance monitor

  • debug radius accounting

  • debug aaa accounting