- Cisco IOS IP SLAs Features Roadmap
- Cisco IOS IP SLAs Overview
- Configuring UDP Jitter Operations
- Configuring UDP Jitter Operations for VoIP
- Configuring a LSP Health Monitor with LSP Discovery
- Configuring IP SLAs for Metro-Ethernet
- Configuring UDP Echo Operations
- Configuring HTTP Operations
- Configuring TCP Connect Operations
- Configuring ICMP Echo Operations
- Configuring ICMP Path Echo Operations
- Configuring ICMP Path Jitter Operations
- Configuring FTP Operations
- Configuring DNS Operations
- Configuring DHCP Operations
- Configuring DLSw+ Operations
- Configuring a Multioperation Scheduler
- Configuring Proactive Threshold Monitoring of IP SLAs Operations
- Finding Feature Information
- Contents
- Restrictions for IP SLAs UDP Jitter Operations for VoIP
- Information About IP SLAs UDP Jitter Operations for VoIP
- How to Configure IP SLAs UDP Jitter Operations for VoIP
Configuring Cisco IOS IP SLAs UDP Jitter Operations for VoIP
This document describes how to configure a Cisco IOS IP Service Level Agreements (SLAs) User Datagram Protocol (UDP jitter operation to proactively monitor Voice over IP (VoIP) quality levels in your network, allowing you to guarantee VoIP quality levels to your users in IPv4 or IPv6 networks. The IP SLAs VoIP UDP jitter operation accurately simulates VoIP traffic using common codecs, and calculates consistent voice quality scores (MOS and ICPIF) between Cisco IOS devices in the network.

Note The term "Voice" in this document should be taken to mean any Internet telephony applications. The term "Voice over IP" can include the transmission of multimedia (both voice and video) over IP networks.
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the "Feature Information for IP SLAs UDP Jitter Operations for VoIP" section.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Contents
•Restrictions for IP SLAs UDP Jitter Operations for VoIP
•Information About IP SLAs UDP Jitter Operations for VoIP
•How to Configure IP SLAs UDP Jitter Operations for VoIP
•Configuration Examples for IP SLAs UDP Jitter Operations for VoIP
•Feature Information for IP SLAs UDP Jitter Operations for VoIP
Restrictions for IP SLAs UDP Jitter Operations for VoIP
•This feature uses UDP traffic to generate approximate Voice over IP scores. It does not provide support for the Real-Time Transport Protocol (RTP).
•ICPIF and MOS values provided by this feature, while consistent within IP SLAs, are estimates only and are intended only for relative comparisons. The values may not match values determined using other methods.
•Predictions of customer opinion (such as those listed for the E-Model transmission rating factor R and derived Mean Opinion Scores) determined by any method are intended only for transmission planning and analysis purposes and should not be interpreted as reflecting actual customer opinions.
Information About IP SLAs UDP Jitter Operations for VoIP
•The Calculated Planning Impairment Factor (ICPIF)
•Voice Performance Monitoring Using IP SLAs
•Codec Simulation Within IP SLAs
The Calculated Planning Impairment Factor (ICPIF)
The ICPIF originated in the 1996 version of ITU-T recommendation G.113, "Transmission impairments," as part of the formula Icpif = Itot - A. ICPIF is actually an acronym for "(Impairment) Calculated Planning Impairment Factor," but should be taken to simply mean the "calculated planning impairment factor." The ICPIF attempts to quantify, for comparison and planning purposes, the key impairments to voice quality that are encountered in the network.
The ICPIF is the sum of measured impairment factors (total impairments, or Itot) minus a user-defined access Advantage Factor (A) that is intended to represent the user's expectations, based on how the call was placed (for example, a mobile call versus a land-line call). In its expanded form, the full formula is expressed as:
Icpif = Io + Iq + Idte + Idd + Ie - A
where
•Io represents impairments caused by non-optimal loudness rating,
•Iq represents impairments caused by PCM quantizing distortion,
•Idte represents impairments caused by talker echo,
•Idd represents impairments caused by one-way transmission times (one-way delay),
•Ie represents impairments caused by equipment effects, such as the type of codec used for the call and packet loss, and
•A represents an access Advantage Factor (also called the user Expectation Factor) that compensates for the fact that users may accept some degradation in quality in return for ease of access.
ICPIF values are expressed in a typical range of 5 (very low impairment) to 55 (very high impairment). ICPIF values numerically less than 20 are generally considered "adequate." While intended to be an objective measure of voice quality, the ICPIF value is also used to predict the subjective effect of combinations of impairments. Table 1, taken from G.113 (02/96), shows how sample ICPIF values are expected to correspond to subjective quality judgement.
For further details on the ICPIF, see the 1996 version of the G.113 specification.

Note The latest version of the ITU-T G.113 Recommendation (2001), no longer includes the ICPIF model. Instead, it refers implementers to G.107: "The Impairment Factor method, used by the E-model of ITU-T G.107, is now recommended. The earlier method that used Quantization Distortion Units is no longer recommended."
The full E-Model (also called the ITU-T Transmission Rating Model), expressed as R = Ro - Is - Id - Ie + A, provides the potential for more accurate measurements of call quality by refining the definitions of impairment factors (see the 2003 version of the G.107 for details). Though the ICPIF shares terms for impairments with the E-Model, the two models should not be confused.
The IP SLAs VoIP UDP Operation feature takes advantage of observed correspondences between the ICPIF, transmission rating factor R, and MOS values, but does not yet support the E-Model.
IP SLAs uses a simplified ICPIF formula, defined in more detail later in this document.
Mean Opinion Scores (MOS)
The quality of transmitted speech is a subjective response of the listener. Each codec used for transmission of Voice over IP provides a certain level of quality. A common benchmark used to determine the quality of sound produced by specific codecs is MOS. With MOS, a wide range of listeners have judged the quality of voice samples sent using particular codecs, on a scale of 1 (poor quality) to 5 (excellent quality). The opinion scores are averaged to provide the mean for each sample. Table 2 shows MOS ratings and the corresponding description of quality for each value.
As the MOS ratings for codecs and other transmission impairments are known, an estimated MOS can be computed and displayed based on measured impairments. This estimated value is designated as MOS-CQE (Mean Opinion Score; Conversational Quality, Estimated) by the ITU in order to distinguish it from objective or subjective MOS values (see P.800.1 for details).
Voice Performance Monitoring Using IP SLAs
One of the key metrics in measuring voice and video quality over an IP network is jitter. Jitter is the name used to indicate the variation in delay between arriving packets (inter-packet delay variance). Jitter affects voice quality by causing uneven gaps in the speech pattern of the person talking. Other key performance parameters for voice and video transmission over IP networks include latency (delay) and packet loss. IP SLAs is an embedded active monitoring feature of Cisco IOS software that provides a means for simulating and measuring these parameters in order to ensure your network is meeting or exceeding service-level agreements with your users.
IP SLAs provides a UDP jitter operation, which consists of UDP probe packets sent across the network from an origin device to a specific destination (called the operational target). This synthetic traffic is used to record the amount of jitter for the connection, as well as the round-trip time, per-direction packet loss, and one-way delay time (one-way latency). (The term "synthetic traffic" indicates that the network traffic is simulated; that is, the traffic is generated by IP SLAs.) Data, in the form of collected statistics, can be displayed for multiple tests over a user-defined period of time, allowing you to see, for example, how the network performs at different times of the day, or over the course of a week. The jitter probe has the advantage of utilizing the IP SLAs Responder to provide minimal latency at the receiving end.
The IP SLAs VoIP UDP jitter operation modifies the standard UDP jitter operation by adding the capability to return MOS and ICPIF scores in the data collected by the operation, in addition to the metrics already gathered by the UDP jitter operation. This VoIP-specific implementation provides even more useful information in determining the performance of your VoIP network, thereby improving your ability to perform network assessment, troubleshooting, and health monitoring.
Codec Simulation Within IP SLAs
The IP SLAs VoIP UDP jitter operation computes statistics by sending n UDP packets, each of size s, sent t milliseconds apart, from a given source router to a given target router, at a given frequency f. The target router must be running the IP SLAs Responder in order to process the probe operations.
To generate MOS and ICPIF scores, you specify the codec type used for the connection when configuring the VoIP UDP jitter operation. Based on the type of codec you configure for the operation, the number of packets (n), the size of each payload (s), the inter-packet time interval (t), and the operational frequency (f) will be auto-configured with default values. (See Table 3 for specifics.) However, you are given the option, if needed, to manually configure these parameters in the syntax of the udp-jitter command.
Table 3 shows the default parameters that are configured for the operation by codec.
For example, if you configure the VoIP UDP jitter operation to use the characteristics for the g711ulaw codec, by default a probe operation will be sent once a minute (f). Each probe operation would consist of 1000 packets (n), with each packet containing 180 bytes of synthetic data (s), sent 20 milliseconds apart (t).
The IP SLAs ICPIF Value
ICPIF value computation with Cisco IOS software is based primarily on the two main factors that can impair voice quality: delayed packets and lost packets. Because packet delay and packet loss can be measured by IP SLAs, the full ICPIF formula, Icpif = Io + Iq + Idte + Idd + Ie - A, is simplified by assuming the values of Io, Iq, and Idte are zero, resulting in the following formula:
Total Impairment Factor (Icpif) = Delay Impairment Factor (Idd) + Equipment Impairment Factor (Ie) - Expectation/Advantage Factor (A)
This means that the ICPIF value is computed by adding a Delay Impairment Factor, which is based on a measurement of delayed packets, and an Equipment Impairment Factor, which is based on a measurement of lost packets. From this sum of the total impairments measured in the network, an impairment variable (the Expectation Factor) is subtracted to yield the ICPIF.
This is the same formula used by Cisco Gateways to calculate the ICPIF for received VoIP data streams.
The Delay Impairment Factor
The Delay Impairment Factor (Idd) is a number based on two values. One value is fixed and is derived using the static values (as defined in the ITU standards) for Codec Delay, Look Ahead Delay, and Digital Signal Processing (DSP) Delay. The second value is variable and is based on the measured one-way delay (round-trip time measurement divided by 2). The one-way delay value is mapped to a number using a mapping table that is based on a G.107 (2002 version) analytic expression. Table 4 shows sample correspondences between the one-way delay measured by IP SLAs and Delay Impairment Factor values.
|
|
---|---|
50 |
1 |
100 |
2 |
150 |
4 |
200 |
7 |
The Equipment Impairment Factor
The Equipment Impairment Factor (Ie) is a number based on the amount of measured packet loss. The amount of measured packet loss, expressed as a percentage of total number of packets sent, corresponds an Equipment Impairment Factor that is defined by codec. Table 5 shows sample correspondences between the packet loss measured by IP SLAs and Equipment Impairment Factor values.
The Expectation Factor
The Expectation Factor, also called the Advantage Factor (A), is intended to represent the fact that users may accept some degradation in quality in return for ease of access. For example, a mobile phone user in a hard-to-reach location may have an expectation that the connection quality will not be as good as a traditional land-line connection. This variable is also called the Advantage Factor (short for Access Advantage Factor) because it attempts to balance an increased access advantage against a decline in voice quality.
Table 6, adapted from ITU-T Rec. G.113, defines a set of provisional maximum values for A in terms of the service provided.
These values are only suggestions. To be meaningful, the use of the factor A and its selected value in a specific application should be used consistently in any planning model you adopt. However, the values in Table 6 should be considered as the absolute upper limits for A.
The default Advantage Factor for IP SLAs VoIP UDP jitter operations is always zero.
The IP SLAs MOS Value
IP SLAs uses an observed correspondence between ICPIF and MOS values to estimate an MOS value. Usage of the abbreviation MOS within the context of this feature should be taken to represent the MOS-CQE (Mean Opinion Score; Conversational Quality, Estimated).
The E model, as defined in G.107 (03/2003), predicts the subjective quality that is experienced by an average listener by combining the impairment caused by transmission parameters (such as loss and delay) into a single rating, the transmission rating factor R (the R Factor). This rating, expressed in a scale of 0 (worst) to 100 (best) can be used to predict subjective user reactions, such as the MOS. Specifically, the MOS can be obtained from the R Factor with a converting formula. Conversely, a modified inverted form can be used to calculate R Factors from MOS values.
There is also a relationship between the ICPIF value and the R Factor. IP SLAs takes advantage of this correspondence by deriving the approximate MOS score from an estimated R Factor, which, in turn, is derived from the ICPIF score. Table 7 shows the resulting MOS values that will be generated for corresponding ICPIF values.
|
|
|
---|---|---|
0 - 3 |
5 |
Best |
4 - 13 |
4 |
High |
14 - 23 |
3 |
Medium |
24 - 33 |
2 |
Low |
34 - 43 |
1 |
Poor |
IP SLAs will always express the estimated MOS value as a number in the range of 1 to 5, with 5 being the best quality. A MOS value of 0 (zero) indicates that MOS data could not be generated for the operation.
How to Configure IP SLAs UDP Jitter Operations for VoIP
•Configuring and Scheduling an IP SLAs VoIP UDP Jitter Operation
Configuring and Scheduling an IP SLAs VoIP UDP Jitter Operation

Note The VoIP-specific implementation of the IP SLAs UDP jitter operation contains different configuration options than the standard UDP jitter operation. As soon as you specify the codec keyword in the udp-jitter command syntax, you are configuring the VoIP-specific implementation of the jitter operation.
Restrictions
•Currently, IP SLAs supports only the following speech codecs (compression methods):
–G.711 A Law (g711alaw: 64 kbps PCM compression method)
–G.711 mu Law (g711ulaw: 64 kbps PCM compression method)
–G.729A (g729a: 8 kbps CS-ACELP compression method)
•The following commands, available in UDP jitter configuration mode, are not valid for UDP jitter (codec) operations:
–history distributions-of-statistics-kept
–history statistics-distribution-interval
–request-data-size
•Specifying the codec-type will configure the appropriate default values for the codec-interval, codec-size, and codec-numpacket options. You should not specify values for the interval, size, and number of packet options unless you have a specific reason to override the defaults (for example, approximating a different codec).
•The show ip sla configuration command will list the values for the "Number of statistic distribution buckets kept" and "Statistic distribution interval (milliseconds)," but these values do not apply to jitter (codec) operations.
SUMMARY STEPS
1. enable
2. configure terminal
3. ip sla operation-number
4. udp-jitter {destination-ip-address | destination-hostname} destination-port codec codec-type [codec-numpackets number-of-packets] [codec-size number-of-bytes] [codec-interval milliseconds] [advantage-factor value] [source-ip {ip-address | hostname}] [source-port port-number] [control {enable | disable}]
5. history enhanced [interval seconds] [buckets number-of-buckets]
6. frequency seconds
7. history hours-of-statistics-kept hours
8. owner owner-id
9. tag text
10. threshold milliseconds
11. timeout milliseconds
12. tos number
or
traffic-class number
13. flow-label number
14. verify-data
15. vrf vrf-name
16. exit
17. ip sla schedule operation-number [life {forever | seconds}] [start-time {hh:mm[:ss] [month day | day month] | pending | now | after hh:mm:ss}] [ageout seconds] [recurring]
18. exit
19. show ip sla configuration [operation-number]
DETAILED STEPS
Troubleshooting Tips
•If the IP SLAs operation is not running and generating statistics, add the verify-data command to the configuration of the operation (while configuring in IP SLA configuration mode) to enable data verification. When enabled, each operation response is checked for corruption. Use the verify-data command with caution during normal operations because it generates unnecessary overhead.
•Use the debug ip sla trace and debug ip sla error commands to help troubleshoot issues with an IP SLAs operation.
What to Do Next
To add proactive threshold conditions and reactive triggering for generating traps, or for starting another operation, to an IP SLAs operation, see Configuring Proactive Threshold Monitoring.
To view and interpret the results of an IP SLAs operation use the show ip sla statistics command. Checking the output for fields that correspond to criteria in your service level agreement will help you determine whether the service metrics are acceptable.
Configuration Examples for IP SLAs UDP Jitter Operations for VoIP
•Example: IP SLAs VoIP UDP Operation Configuration
•Example: IP SLAs VoIP UDP Operation Statistics Output
Example: IP SLAs VoIP UDP Operation Configuration
The following example assumes that the IP SLAs Responder is enabled on the device at 209.165.200.225.
Router> enable
Password:
Router# configure terminal
Enter configuration commands, one per line. End with the end command.
Router(config)# ip sla 10
Router(config-sla)# udp-jitter 209.165.200.225 16384 codec g711alaw advantage-factor 2
Router(config-sla-jitter)# owner admin_bofh
Router(config-sla-jitter)# exit
Router(config)# ip sla schedule 10 start-time now
Router(config)# exit
Router#
Router# show running-config | begin ip sla 10
ip sla 10
udp-jitter 209.165.200.225 16384 codec g711alaw advantage-factor 2
owner admin_bofh
ip sla schedule 10 start-time now
.
.
.
Router# show ip sla configuration 10
Entry number: 10
Owner: admin_bofh
Tag:
Type of operation to perform: jitter
Target address: 209.165.200.225
Source address: 0.0.0.0
Target port: 16384
Source port: 0
Operation timeout (milliseconds): 5000
Codec Type: g711alaw
Codec Number Of Packets: 1000
Codec Packet Size: 172
Codec Interval (milliseconds): 20
Advantage Factor: 2
Type Of Service parameters: 0x0
Verify data: No
Vrf Name:
Control Packets: enabled
Operation frequency (seconds): 60
Next Scheduled Start Time: Start Time already passed
Life (seconds): 3600
Entry Ageout (seconds): never
Status of entry (SNMP RowStatus): Active
Connection loss reaction enabled: No
Timeout reaction enabled: No
Verify error enabled: No
Threshold reaction type: Never
Threshold (milliseconds): 5000
Threshold Falling (milliseconds): 3000
Threshold Count: 5
Threshold Count2: 5
Reaction Type: None
Number of statistic hours kept: 2
Number of statistic distribution buckets kept: 1
Statistic distribution interval (milliseconds): 20
Enhanced History:
When a codec type is configured for a jitter operation, the standard jitter "Request size (ARR data portion)," "Number of packets," and "Interval (milliseconds)" parameters will not be displayed in the show ip sla configuration command output. Instead, values for "Codec Packet Size," "Codec Number of Packets," and "Codec Interval (milliseconds)" are displayed.
Example: IP SLAs VoIP UDP Operation Statistics Output
Use the show ip sla statistics command to display Voice scores (ICPIF and MOS values) for the jitter (codec) operation.
Router# show ip sla statistics 10
Entry number: 10
Modification time: 12:57:45.690 UTC Sun Oct 26 2003
Number of operations attempted: 1
Number of operations skipped: 0
Current seconds left in Life: Forever
Operational state of entry: Active
Last time this entry was reset: Never
Connection loss occurred: FALSE
Timeout occurred: FALSE
Over thresholds occurred: FALSE
Latest RTT (milliseconds): 19
Latest operation start time: 12:57:45.723 Sun Oct 26 2003
Latest operation return code: OK
!
Voice Scores:
ICPIF: 20 MOS Score: 3.20
!
RTT Values:
NumOfRTT: 10 RTTAvg: 19 RTTMin: 19 RTTMax: 20
RTTSum: 191 RTTSum2: 3649
Packet Loss Values:
PacketLossSD: 0 PacketLossDS: 0
PacketOutOfSequence: 0 PacketMIA: 0 PacketLateArrival: 0
InternalError: 0 Busies: 0
Jitter Values:
NumOfJitterSamples: 9
MinOfPositivesSD: 0 MaxOfPositivesSD: 0
NumOfPositivesSD: 0 SumOfPositivesSD: 0 Sum2PositivesSD: 0
MinOfNegativesSD: 0 MaxOfNegativesSD: 0
NumOfNegativesSD: 0 SumOfNegativesSD: 0 Sum2NegativesSD: 0
MinOfPositivesDS: 1 MaxOfPositivesDS: 1
NumOfPositivesDS: 1 SumOfPositivesDS: 1 Sum2PositivesDS: 1
MinOfNegativesDS: 1 MaxOfNegativesDS: 1
NumOfNegativesDS: 1 SumOfNegativesDS: 1 Sum2NegativesDS: 1
Interarrival jitterout: 0 Interarrival jitterin: 0
One Way Values:
NumOfOW: 0
OWMinSD: 0 OWMaxSD: 0 OWSumSD: 0 OWSum2SD: 0
OWMinDS: 0 OWMaxDS: 0 OWSumDS: 0 OWSum2DS: 0
Additional References
Related Documents
|
|
---|---|
Cisco IOS commands |
|
Cisco IOS IP SLAs commands |
|
Voice over IP (VoIP) codecs |
"Understanding Codecs: Complexity, Hardware Support, MOS, and Negotiation" http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml |
Jitter in Packet Voice Networks |
"Understanding Jitter in Packet Voice Networks (Cisco IOS Platforms)" http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800945df.shtml |
PSTN Fallback for Voice Gateways |
"SIP: Measurement-Based Call Admission Control for SIP" http://www.cisco.com/en/US/docs/ios/12_2t/12_2t15/feature/guide/ftcacsip.html |
Standards
|
|
---|---|
ITU-T Recommendation G.107 (2003) |
The E-model, a computation model for use in transmission planning |
ITU-T Recommendation G.113 (1996) |
Transmission impairments |
ITU-T Recommendation G.113 (2001) |
Transmission impairments due to speech processing |
ITU-T Recommendation G.711 (1998) |
Pulse code modulation (PCM) of voice frequencies |
ITU-T Recommendation G.729 Annex A (1996) |
Reduced complexity 8 kbit/s CS-ACELP speech codec |
ITU-T Recommendation P.800.1 (2003) |
Mean Opinion Score (MOS) terminology |
1 Full support by this feature for listed RFCs is not claimed. ITU Telecommunication Standards ("ITU-T Recommendations In Force") can be obtained from http://www.itu.ch. Summary definitions are available from a variety of internet sources. |
MIBs
|
|
---|---|
CISCO-RTTMON-MIB |
To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL: |
RFCs
|
|
---|---|
RFC 768 |
User Datagram Protocol |
RFC 1889 |
RTP: A Transport Protocol for Real-Time Applications |
1 Full support by this feature for listed RFCs is not claimed. |
Technical Assistance
Feature Information for IP SLAs UDP Jitter Operations for VoIP
Table 8 lists the features in this module and provides links to specific configuration information.
Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 8 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.
Glossary
codec—In the context of IP Telephony, a codec is a compression and decompression algorithm used to transfer voice and video data more efficiently. Voice codec types are typically referred to using the ITU recommendation number that defines the algorithm (for example, "G.711" instead of "PCM").
CS-ACELP—The codec type defined in the reference documents G.729 and G.729A, Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP).
ITU—The International Telecommunication Union. The ITU is an international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services. The ITU Telecommunication Standardization Sector (ITU-T), responsible for defining standards (Recommendations) covering all fields of telecommunications, is one of the three operational sectors of the ITU. The ITU web site is at http://www.itu.int.
ITU-T—ITU Telecommunication Standardization Sector. The ITU-T is one of the three operational sectors of the ITU, and is responsible for defining standards (called ITU-T Recommendations) covering all fields of telecommunications.
MOS-CQE (Mean Opinion Score; Conversational Quality, Estimated)—The score calculated by a network planning model which aims at predicting the quality in a conversational application situation. Estimates of conversational quality carried out according to ITU-T Rec. G.107, when transformed to a mean opinion score (MOS), give results in terms of MOS-CQE.1
PCM—The codec type defined in the reference document G.711, Pulse code modulation (PCM) of voice frequencies.