Table Of Contents
Call Admission Control for H.323 VoIP Gateways
User Selected Call Admission Controls
Resource Unavailable Signaling
Supported Standards, MIBs, and RFCs
Configuring Call Threshold Poll Interval
Verifying Call Admission Control Tasks
Call Spike Configuration Example
Call Threshold Configuration Example
Call Threshold Poll Interval Configuration Example
Call Treatment Configuration Example
Call Admission Control for H.323 VoIP Gateways
Document Update Alert
This document was originally produced for Cisco IOS Release 12.2(11)T. This feature has been updated in subsequent releases, and more recent documentation is available.
If you are using Cisco IOS Release 12.2(11)T or higher, refer to the following documentation in the Cisco IOS Voice Configuration Library, Release 12.3:
•
Trunk Connections and Conditioning Features
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/vcltrunk.htm
Feature History
This document describes Call Admission Control for H.323 VoIP Gateways feature. The PSTN Fallback feature is an additional enhancement. This document includes the following sections:
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Supported Standards, MIBs, and RFCs
Feature Overview
This feature provides the ability to support resource-based call admission control processes. These resources include system resources such as CPU, memory, and call volume, and interface resources such as call volume.
If system resources are not available to admit the call, two kinds of actions are provided: system denial (which busies out all of T1 or E1) or per call denial (which disconnects, hairpins, or plays a message or tone). If the interface-based resource is not available to admit the call, the call is dropped from the session protocol (such as H.323).
User Selected Call Admission Controls
The Call Admission Control for H.323 VoIP Gateways feature allows a user to configure thresholds for local resources as well as memory and CPU resources. The list of local resources that are configured for call admission are described on the call threshold poll-interval command reference page.
With the call threshold command, a user is allowed to configure two thresholds, high and low, for each resource. Call treatment is triggered when the current value of a resource goes beyond the configured high. The call treatment remains in effect until the current resource value falls below the configured low. Having high and low thresholds prevents call admission flapping and provides hysteresis in call admission decision making.
With the call spike command, a user is allowed to configure the limit for incoming calls during a specified time period. A call spike is the term for when a large number of incoming calls arrive from the Public Switched Telephone Network (PSTN) in a very short period of time (for example:100 incoming calls in 10 milliseconds).
With the call treatment command, users are allowed to select how the call should be treated when local resources are not available to handle the call. For example, when the current resource value for any one of the configured triggers for call threshold has exceeded the configured threshold, the call treatment choices are as follows:
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Time-division multiplexing (TDM) hairpinning—Hairpins the calls through the plain old telephone service (POTS) dial peer.
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Reject—Disconnects the call.
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Play message or tone—Plays a configured message or tone to the user.
Resource Unavailable Signaling
The Resource Unavailable Signaling feature supports the autobusyout feature where channels are busied out when local resources are not available to handle the call. Autobusyout is supported on both channel associated signaling (CAS) and Primary Rate Interface (PRI) channels.
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CAS—Uses busyout to signal that local resources are unavailable.
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PRI—Uses either service messages or a cause code to signal that resources are unavailable.
PSTN Fallback
The goal of PSTN fallback is to monitor congestion in the IP network and either redirect calls to the PSTN or reject calls based on the network congestion. Calls can be rerouted to an alternate IP destination or to the PSTN if the IP network is found unsuitable for voice traffic at that time. The user defines the congestion thresholds based on the configured network. This functionality enables the service provider to give a reasonable guarantee about the quality of the conversation to their Voice over IP (VoIP) users at the time of call admission.
Note
PSTN fallback does not provide assurances that a VoIP call that proceeds over the IP network is protected from the effects of congestion. This is the function of the other quality of service (QoS) mechanisms such as IP Real-Time Transport Protocol (RTP) priority or low latency queueing (LLQ).
PSTN fallback includes the following features:
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Offers flexibility to define the congestion thresholds based on the network.
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Defines a threshold based on Calculated Planning Impairment Factor (ICPIF), which is derived as part of International Telecommunication Union (ITU) G.113.
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Defines a threshold based solely on packet delay and loss measurements.
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Uses Service Assurance Agent (SAA) probes to provide packet delay, jitter, and loss information for the relevant IP addresses. Based on the packet loss, delay, and jitter encountered by these probes, an ICPIF or delay and loss values are calculated.
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Is supported by calls of any codec. Only G.729 and G.711 have accurately simulated probes. Calls of all other codecs are emulated by a G.711 probe.
For more information, including configuration tasks and examples, and command reference pages for PSTN fallback, refer to the Cisco IOS Voice, Video, and Fax Configuration Guide and Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2.
Benefits
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Configurable call treatment — Allows an Internet service provider (ISP) to configure how the call is supposed to be treated when local resources to process the call are not available.
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TDM hairpinning—Hairpins the calls through the POTS dial peer.
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Reject—Disconnects the call.
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Play message—Plays a configured tone to the user.
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Resource unavailable signaling—Allows user to automatically busy out channels when local resources are not available to handle the call.
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CAS—Uses busyout to signal that resources are unavailable.
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PRI—Uses either service messages or disconnects with correct cause code to signal that resources are unavailable.
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User selected threshold—Allows user to configure thresholds for each of the local resources.
PSTN Fallback
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The PSTN Fallback feature automatically routes a call to any alternate destination when the data network is congested at the time of the call setup.
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PSTN fallback provides delay, jitter, and packet loss information for the configured IP addresses.
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PSTN fallback contains a network traffic cache used to maintain ICPIF and delay, loss, and jitter values, which increases performance. A new call does not have to wait for probe results before it is admitted. The value is cached from a previous call.
Restrictions
The following are restrictions applicable to the PSTN Fallback feature only:
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Upon detecting network congestion, the PSTN Fallback feature does not do anything to the existing call. It affects only subsequent calls.
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There is a single ICPIF/delay-loss value per system.
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The PSTN Fallback feature adds a small call setup delay for the first call to a new IP destination.
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H.323 VoIP calls are supported.
Related Documents
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Single and High-Density VoIP Support for the Cisco AS5300/Voice Gateway
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Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2
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Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2
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Cisco AS5300 Software Configuration Guide
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Advanced Voice Busyout, Cisco IOS Release 12.2(4)T.
Supported Platforms
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Cisco 1750
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Cisco 1751
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Cisco 2600 series
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Cisco 3600 series
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Cisco AS5300
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Cisco AS5350
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Cisco AS5400
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Cisco AS5800
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CiscoAS5850
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Cisco 7200 series
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Cisco MC3810
Determining Platform Support Through Cisco Feature Navigator
Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.
Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.
Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
Note
Cisco Feature Navigator does not support Cisco IOS Release 12.2(2)XA or Release 12.2(2)XB1.
Supported Standards, MIBs, and RFCs
Standards
No new or modified standards are supported by this feature.
MIBs
No new or modified MIBs are supported by this feature.
To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml
RFCs
No new or modified RFCs are supported by this feature.
Prerequisites
The Cisco AS5350 and Cisco AS5400 do not support the MICA technologies modem card, the Microcom modem card, or the VoIP feature card. Voice and modem functions are provided by the Universal Port Dial Feature Card running SPE firmware. For more information, refer to the Cisco AS5350 Universal Gateway Card Installation Guide and the Cisco AS5400 Universal Gateway Card Installation Guide. All references to the Cisco AS5300 in this document apply to the Cisco AS5350 and Cisco AS5400 platforms with the following exceptions:
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Use the Universal Port Dial Feature Card instead of the MICA or Microcom modem cards.
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Use SPE firmware instead of portware version 6.7.7.
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Run Cisco IOS Release 12.1(5)XM2 or higher software for VoIP functionality.
Other Prerequisites
Before you configure PSTN fallback, you must have already configured VoIP. For more information, refer to the Cisco IOS Voice, Video, and Fax Configuration Guide and Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2.
Use Table 1 to ensure that you have the correct Cisco IOS release for your platform.
Configuration Tasks
See the following sections for the configuration tasks for the Call Admission Control and PSTN Fallback features. Each task in the list is identified as either required or optional:
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Configuring Call Spike (required)
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Configuring Call Threshold (required)
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Configuring Call Threshold Poll Interval (optional)
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Configuring Call Treatment (optional)
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Configuring PSTN Fallback (required)
Configuring Call Spike
To configure the limit for the number of incoming calls in a period of time, enter the following command in global configuration mode:
Command PurposeRouter(config)# call spike call-number [steps number-of-steps size milliseconds]
Configures the limit for the number of incoming calls in a short period of time.
Configuring Call Threshold
To configure the call threshold, use the following command in global configuration mode:
Configuring Call Threshold Poll Interval
To configure the interval at which the call threshold is polled, use the following command in global configuration mode:
Command PurposeRouter(config)# call threshold poll-interval {cpu-average | memory} seconds
Enables a polling interval threshold for CPU or memory.
Configuring Call Treatment
To configure the call treatment, use the following command in global configuration mode:
Verifying Call Admission Control Tasks
To verify the call admission control, use the following commands:
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show call spike status—Displays the configured call spike threshold and statistics for incoming calls.
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show call threshold—Displays enabled triggers, current values for configured triggers, and number of Application Programming Interface (API) calls that were made to global and interface resources.
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show call treatment—Displays the call treatment configuration and the statistics for handling the calls based upon resource availability.
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show running-config—Displays the configuration of all of the configuration tasks commands.
Configuring PSTN Fallback
For configuration information, refer to the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2.
Configuration Examples
This section provides the following configuration examples:
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Call Spike Configuration Example
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Call Threshold Configuration Example
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Call Threshold Poll Interval Configuration Example
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Call Treatment Configuration Example
Call Spike Configuration Example
The following configuration of the call spike command has a call number of 30, 10 steps, and a step size of 2,000 milliseconds:
call threshold global cpu-avg low 70 high 80call spike 30 steps 10 size 2000cns event-service serverCall Threshold Configuration Example
The following example will busy out the total-calls resource of 5 (low) or 5,000 (high):
call threshold global total-calls low 5 high 5000 busyoutThe following example enables thresholds of 5 (low) and 2,500 (high) on interface Ethernet 0:
call threshold interface Ethernet 0 int-calls low 5 high 2500The following example will busyout the average CPU utilization if 5 percent (low) or 65 percent (high) is reached:
call threshold global cpu-avg low 5 high 65 busyoutCall Threshold Poll Interval Configuration Example
The following example enables a polling interval threshold for memory of 10 seconds:
call threshold poll-interval memory 10The following example enables a polling interval threshold of 50 seconds:
call threshold poll-interval cpu-average 50Call Treatment Configuration Example
The following example enables the Call Treatment feature with a "hairpin" action:
call treatment oncall treatment action hairpinThe following example displays proper formatting of the action playmsg keywords:
call treatment oncall treatment action playmsg tftp://keyer/prompts/conjestion.au
Note
The congestion.au file plays when local resources are not available to handle the call.
The following example configures a call treatment cause code to display no-qos when local resources are unavailable to process a call:
call treatment oncall treatment cause-code no-qosCommand Reference
This section documents modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.2 command reference publications.
call spike
To configure the limit of incoming calls in a short period of time, use the call spike command in global configuration mode. To disable this command, use the no form of this command.
call spike call-number [steps number-of-steps size milliseconds]
no call spike
Syntax Description
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Usage Guidelines
A call spike is the term for a large number of incoming calls arriving from the PSTN in a very short period of time (for example:100 incoming calls in 10 milliseconds). This command allows you to control the amount of call requests that are received in a configured time period.
Examples
The following configuration of the call spike command has a call number of 30, 10 steps, and a step size of 2,000 milliseconds:
call spike 30 steps 10 size 2000Related Commands!
Command Descriptionshow call spike status
Displays the configuration of the threshold for incoming calls.
call threshold
To enable the global resources of this gateway, use the call threshold command in global configuration mode. To disable this command, use the no form of this command.
call threshold {global trigger-name | interface interface-name interface-number int-calls} low value high value [busyout | treatment]
no call threshold {global trigger-name | interface interface-name int-calls}
Syntax Description
Defaults
The defaults for the busyout and treatment keywords are for global resource triggers only.There are no other defaults.
Command Modes
Global configuration
Command History
Examples
The following example will busyout the total-calls resource of 5 (low) or 5,000 (high) is reached:
call threshold global total-calls low 5 high 5000 busyoutThe following example enables thresholds of 5 (low) and 2,500 (high) for interface calls on interface Ethernet 0:
call threshold interface Ethernet 0 int-calls low 5 high 2500The following example will busyout the average CPU utilization if 5 percent (low) or 65 percent (high) is reached:
call threshold global cpu-avg low 5 high 65 busyoutRelated Commands
call threshold poll-interval
To enable a polling interval threshold for CPU or memory, use the call threshold poll-interval command in global configuration command. To disable this command, use the no form of this command.
call threshold poll-interval {cpu-average | memory} number-of-seconds
no call threshold poll-interval {cpu-average | memory}
Syntax Description
Defaults
The default value for the cpu-average keyword is 60 seconds.The default value for the memory keyword is 5 seconds.
Command Modes
Global configuration
Command History
Examples
The following example enables a polling interval threshold for memory of 10 seconds:
call threshold poll-interval memory 10Related Commands
call treatment
To configure how calls should be processed when local resources are unavailable, use the call treatment command in global configuration mode. To disable the call treatment triggers, use the no form of this command.
call treatment {on | action action [value] | cause-code cause-code | isdn-reject value}
no call treatment {on | action action [value] | cause-code cause-code | isdn-reject value}
Syntax Description
Defaults
The treatment is inactive by default.
Command Modes
Global configuration
Command History
Usage Guidelines
This command indicates whether the call is to be disconnected with a cause code, hairpinned, message played to the user, or busy tone to the user.
Examples
The following example enables the Call Treatment feature with a `hairpin' action:
call treatment oncall treatment action hairpinThe following example displays proper formatting of the playmsg action keyword:
call treatment action playmsg tftp://keyer/prompts/conjestion.au
Note
Thecongestion.au file playswhen local resources are not available to handle the call.
The following example configures a call treatment cause-code to display no-Qos when local resources are unavailable to process a call:
call treatment cause-code no-qosRelated Commands
clear call threshold
To clear enabled triggers and their associated parameters, use the clear call threshold command in EXEC mode.
clear call threshold {stats | total-calls [value] | interface int-name int-calls [value]
Syntax Description
Defaults
The default for the total-calls keyword and the value argument is 0.
Command Modes
EXEC mode
Command History
Examples
The following example resets all call threshold stats:
clear call threshold statsThe following example resets the counter for the call volume in the gateway:
clear call threshold total-callsThe following example resets the counter for the call volume on interface Ethernet 0:
clear call threshold interface e 0 int-callsRelated Commands
clear call treatment stats
To clear the call treatment stats, use the clear call treatment stats command in EXEC mode.
clear call treatment stats
Syntax Description
There are no keywords for this command.
Defaults
No default behavior or values.
Command Modes
EXEC mode
Command History
Examples
See the following sample output from the clear call treatment stats command in EXEC mode:
Router# clear call treatment statsRelated Commands
debug call threshold
To see details of the trigger actions, use the debug call threshold command in privileged EXEC command. To disable debugging output, use the no form of this command.
debug call threshold module
no debug call threshold
Syntax Description
module
The module argument can be:
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core—Traces the resource information.
•
detail—Traces for detail information.
Defaults
Debugging is disabled.
Command Modes
EXEC mode
Command History
Examples
Router# debug call threshold coreRSCCAC Core info debugging is onRouter# debug call threshold detailAll RSCCAC info debugging is ondebug call treatment action
To debug the call treatment actions, use the debug call treatment action command in privileged EXEC command. Use the no form of this command to disable debugging output. .
debug call treatment action
no debug call treatment action
Syntax Description
This command has no arguments or keywords.
Defaults
Debugging is disabled.
Command Modes
EXEC
Command History
Examples
Debug actions are performed on calls by call treatment. The following sample output shows that call treatment is turned on:
Router# debug call treatment actionCall treatment action debugging is onds0-group
To define T1or E1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, enter the ds0-group controller configuration command. To remove the group, signaling, and direction settings, use the no form of the command.
ds0-group ds0-group-number timeslots timeslot-list [service {data | fax | voice} | type {e&m-fgb
| e&m-fgd | e&m-immediate-start | fgd-eana | fgd-os | fxs-ground-start | fxs-loop-start |
none | r1-itu | r1-modified | r1-turkey | sas-ground-start | sas-loop-start}]no ds0-group ds0-group-number
Syntax Description
Defaults
There is no DS-0 group. Calls are allowed in both directions by default.
Command Modes
Controller configuration
Command History
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows on Cisco AS5300 with a T1 controller: slot/port. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Examples
The following example configures ranges of T1 controller timeslots for FXS ground-start signaling
controller T1 1/0ds0-group 1 timeslots 1-4 type fxs-ground-startRelated Commands
show call spike status
To display the configured call spike threshold and statistics for incoming calls, use the show call spike status command in EXEC mode.
show call spike status
Syntax Description
There are no keywords or descriptions for this command.
Defaults
No default behavior or values.
Command Modes
EXEC mode
Command History
Examples
Router# show call spike statusCall Spiking:ConfiguredCall spiking:NOT TRIGGEREDtotal call count in sliding window::20
Related Commands
Command Descriptioncall spike
Configures the limit for the number of incoming calls in a short period of time.
show call threshold
To display enabled triggers, current values for configured triggers, and number of Application Programming Interface (API) calls that were made to global and interface resources, use the show call threshold command in EXEC mode.
show call threshold {configuration | status [unavailable] | stats}
Syntax Description
Defaults
No default behavior or values.
Command Modes
EXEC mode
Command History
Examples
Router# show call threshold configSome resource polling interval:CPU_AVG interval: 60Memory interval: 5IF Type Value Low High Enable----- ---- ----- ---- ---- ------Serial3/1:23 int-calls 0 107 107 N/AN/A cpu-avg 0 70 90 busy&treat
Related Commands
show call treatment
To display the call treatment configuration and the statistics for handling the calls based upon resource availability, use the show call treatment command in EXEC mode.
show call treatment {config | stats}
Syntax Description
config
Displays the call treatment configuration.
stats
Displays the statistics for handling the calls due to resource availability.
Defaults
No default behavior or values
Command Modes
EXEC mode
Command History
Examples
Router# show call treat configCall Treatment Config---------------------Call treatment is OFF.Call treatment action is: RejectCall treatment disconnect cause is: no-resourceCall treatment ISDN reject cause-code is: 41
Router# show call treat statsCall Treatment Statistics-------------------------Total Calls by call treatment: 0Calls accepted by call treatment: 0Calls rejected by call treatment: 0Reason Num. of calls rejected------ ----------------------cpu-5sec: 0cpu-avg: 0total-mem: 0io-mem: 0proc-mem: 0total-calls: 0
Related Commands
Command Descriptioncall treatment
Configures how calls should be processed when local resources are unavailable.
clear call treatment stats
Clears the call treatment stats.
test call threshold
To test how the core Application Programming Interfaces (APIs) behave based on the resource configuration, use the test call threshold command in EXEC mode.
test call threshold {enable [busyout | treatment] [global | ipaddress ipaddress] |
interface interface-name interface-number}Syntax Description
Defaults
No default behavior or values.
Command Modes
EXEC mode
Command History
Examples
The following example specifies the test to be on the global resources.
Router# test call threshold enable globalRelated Commands
Glossary
ABCD signaling—Four-bit telephony line signaling coding in which each letter of "ABCD" represents one of the four bits. This is often associated with CAS or Robbed-Bit signaling on a T1 or E1 telephony trunk.
AIS—Alarm indication signal. In a T1 transmission, an all-ones signal transmitted in lieu of the normal signal to maintain transmission continuity and to indicate to the receiving terminal that there is a transmission fault that is located either at, or upstream from, the transmitting terminal
API—Application Programming Interface. Specification of function-call conventions that defines an interface to a service.
AVBO—Advanced Voice Busy Out.
Cisco-trunk (private line) call—A Cisco-trunk (private line) call is established by the forced connection of a dynamic switched call. A Cisco-trunk call is established during configuration of the trunk and stays up for the duration of the configuration. Optionally, it provides a pass-through connection path to pass signaling information between the two telephony interfaces at either end of the connection.
CLI—Command line interface. Interface that allows the user to interact with the operating system by entering commands and optional arguments.
CODEC—Coder-Decoder. An integrated circuit device that typically uses pulse code modulation to transform analog signals into a digital bit stream and digital signals back into analog signals. In Voice over IP, Voice over Frame Relay, and Voice over ATM, a DSP software algorithm used to compress/decompress speech or audio signals.
DLCI—Data-link connection identifier. Value that specifies a PVC or SVC in a Frame Relay network. In the basic Frame Relay specification, DLCIs are locally significant (connected devices might use different values to specify the same connection). In the LMI extended specification, DLCIs are globally significant (DLCIs specify individual end devices).
Dial peer—An addressable call endpoint that contains configuration information including voice protocol, a CODEC type, and a telephone number associated with the call endpoint. There are five kinds of dial peers: POTS, VoIP, VoFR, VoATM, and VoHDLC.
DS0—Digital signal level 0. Framing specification used in transmitting digital signals over a single channel at 64-kbps on a T1 facility.
DSP—Digital Signal Processor. A specialized computer chip designed to perform speedy and complex operations on digitized waveforms.
DTMF—Dual tone multifrequency. Uses two simultaneous voice-band tones for dial such as touch tone.
DTMF relay—Enables the generation of FRF.11 Annex A frames for a VoFR dial peer. The DSP generates Annex A frames instead of passing a DTMF tone through the network as a voice sample.
Dynamic switched call—A telephone call dynamically established across a packet data network based on a dialed telephone number. In the case of VoFR, a Cisco proprietary session protocol similar to Q.931 is used to achieve call switching and negotiation between calling endpoints. The proprietary session protocol runs over FRF.11-compliant subchannels.
E&M—Stands for recEive and transMit (or Ear and Mouth). E&M is a trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco's analog E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines). E&M is also available on E1 and T1 digital interfaces.
E1—European equivalent of T1. 32-64kbps channels include 1-channel for framing and 1-channel for D-channel information at a 2.048 Mhz clock rate.
FRF—Frame Relay Forum. An association of corporate members consisting of vendors, carriers, users, and consultants committed to implementing Frame Relay in accordance with national and international standards. See http://www.frforum.com.
FXS—Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.
ICPIF—Calculated Planning Impairment Factor. Calculated and used as per the ITU G.113 specification.
IO—Input/output.
LLQ—Low latency queuing. LLQ brings strict priority queueing to Class-Based Weighted Fair Queueing (CBWFQ). Strict priority queueing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic.
MD5—Message Digest 5. Algorithm used for message authentication in SNMP v.2. MD5 verifies the integrity of the communication, authenticates the origin, and checks for timeliness.
MEL CAS—Mercury Exchange Limited (MEL) Channel Associated Signaling. A voice signaling protocol used primarily in the United Kingdom.
OOS—Out of service state of the call or trunk.
PBX—Private Branch Exchange. A privately owned central switching office.
Permanent calls—Permanent calls are private line calls used for fixed point-to-point calls, connections between PBXs (E&M to E&M), or for remote telephone extensions (FXO to FXS).
POTS—Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN.
POTS dial peer—Dial peer connected by a traditional telephony network. POTS peers point to a particular voice port on a voice network device.
PRI—Primary Rate Interface. ISDN interface to primary rate access. Primary rate access consists of a single 64-Kbps D channel plus 23 (T1) or 30 (E1) B channels for voice or data.
PSTN—Public Switched Telephone Network. PSTN refers to the local telephone company.
RAI—Resource Availability Indicator.
SAA—Service Assurance Agent, formerly known as Response Time Reporter (RTR). Works alongside TCP to carry streaming data over the network. RTP uses packet headers that contain sequencing information, time stamps required to time the output (for example, display of frames) and synchronize different data streams (for example, audio and video), and information on the packet's "payload" (for example, MPEG versus H.261 encoding). This payload descriptor allows RTP to support multiple compression types.
Switched calls—Switched calls are normal telephone calls when a user picks up a phone, hears a dial tone and enters the destination phone number to reach the other phone. Switched calls can also be private line auto-ringdown (PLAR) calls, or tie-line calls for fixed point-to-point connections.
T1—Digital WAN carrier facility. T1 transmits DS-1-formatted data at 1.544 Mbps through the telephone-switching network by using AMI or B8ZS coding.
Trunk—Service that allows quasi-transparent connections between two PBXs, a PBX and a local extension, or some other combination of telephony interfaces with signaling passed transparently through the packet data network.
Voice over Frame Relay—Voice over Frame Relay enables a router to carry voice traffic (for example, telephone calls and faxes over a Frame Relay network. When sending voice traffic over Frame Relay, the voice traffic is segmented and encapsulated for transit across the Frame Relay network by using FRF.12 encapsulation.
Voice over IP—Voice over IP enables a router to carry voice traffic, for example, telephone calls and faxes) over an IP network. In Voice over IP, the DSP segments the voice signal into frames, which are then coupled in groups of two and stored in voice packets that are transported by using IP in compliance with ITU-T specification H.323.
Note
For a list of other internetworking terms, see Internetworking Terms and Acronyms, available on the Documentation CD-ROM and Cisco Connection Online (CCO) at the following URL: http://www.cisco.com/univercd/cc/td/doc/cisintwk/ita/index.htm.
