The Bandwidth-Based Call Admission Control (CAC) feature provides the functionality to reject SIP calls when the bandwidth accounted by the SIP signaling layer exceeds the aggregate bandwidth threshold for VoIP media traffic--voice, video, and fax. This functionality helps you prevent Quality of Service (QoS) degradation of VoIP media traffic for existing calls when the bandwidth allocated for VoIP traffic is fully utilized. The Bandwidth-Based Call Admission Control feature is supported on Session Initiation Protocol (SIP) trunks of the Time Division Multiplexing (TDM) SIP gateway and the Cisco Unified Border Element (Cisco UBE).
Midcall media renegotiation can also be rejected if the configured maximum bandwidth threshold for the VoIP media traffic is exceeded. The call continues as per the previously negotiated media codecs if midcall media renegotiation is rejected.
The excess subscription of the bandwidth allocated for VoIP traffic results in VoIP media packets being dropped or delayed, irrespective of the VoIP call to which they belong. Under such circumstances, it is better to deny new calls to prevent QoS deterioration for existing VoIP call traffic. The existing traffic congestion resolution mechanisms do not differentiate between media packets of existing calls (admitted) and new calls (oversubscribed). Similarly, existing call signaling is unaware of the media traffic congestion. The Bandwidth-Based Call Admission Control feature fills this gap by rejecting new SIP calls when the bandwidth allocated for VoIP traffic is fully utilized. The actual bandwidth usage is not measured and policed. The lower-level QoS policies control the traffic characteristics for the specified traffic class.
Note
The Bandwidth-Based Call Admission Control feature is applicable only to VoIP traffic.
Your software release may not support all the features documented in this module. For the latest caveats and feature information, see
Bug Search Tool and 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 table at the end of this module.
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.
Restrictions for Bandwidth-Based Call Admission Control
Cisco UBE, configured with the Bandwidth-Based Call Admission Control feature, will not reject the call if the bandwidth of the SDP answer is greater than the bandwidth of the SDP offer.
Layer 2 overhead is not included in the bandwidth calculation.
A midcall delayed-offer (DO) to DO call is disconnected if the bandwidth requested in an offer message (200 OK) exceeds the threshold bandwidth.
Real Time Transport Control Protocol (RTCP) and RTP Named Telephone Event (RTP-NTE) bandwidth requirement is not computed.
The Bandwidth-Based Call Admission Control feature does not support:
Cisco fax relay.
Filtering of codecs to accommodate calls within the available bandwidth.
Media flow-around, Session Description Protocol (SDP) pass-through, out-of-box low-density transcoding, high-density transcoding, video transcoding, and midcall consumption functionalities.
The bandwidth requirement for each SIP call leg is calculated using the codec information available in the SDP. Here, the actual media bandwidth used is not measured.
Bandwidth in Kbps (Kilo bits per second) = [codec bytes + RTP header (12) + UDP (8) + IP Header (20 or 40)] * Packets per seconds * 8/1000
Where, codec bytes = Codec payload size, in bytes, for a given packetization interval.
RTP header = Size of the RTP header, in bytes.
UDP = Size of the UDP header, in bytes.
IP Header = Size of the IP header, in bytes. The IPV4 header is 20 bytes and the IPV6 header is 40 bytes.
Packets per second = Number of RTP packets sent or received per second. This value is as per the negotiated packetization interval. The SDP media attribute "ptime" indicates the number of packets per second.
Bandwidth Tables
This section provides the sample maximum bandwidth calculation for audio and fax calls.
Table 1
Audio Bandwidth Table
Codec and Bit Rate (Kbps)
Codec Sample Size in Bytes
Voice Payload Size in Bytes
Voice Payload Size in Milliseconds
Packets Per Second
Bandwidth for IPv4 (excluding Layer 2) in Kbps
Bandwidth for IPv6 (excluding Layer 2) in Kbps
G.711 (64 Kbps)
80
160
20
50
80
88
G.729 (8 Kbps)
10
20
20
50
24
32
G.723.1 (6.3 Kbps)
24
24
30
33.3
17
22
G.723.1 (5.3 Kbps)
20
20
30
33.3
16
21
G.726 (32 Kbps)
20
80
20
50
48
56
G.726 (24 Kbps)
15
60
20
50
40
48
G.726 (16 Kbps)
10
40
20
50
32
40
G.728 (16 Kbps)
10
40
20
50
32
40
G722_64k (64 Kbps)
80
160
20
50
80
88
ilbc_mode_20 (15.2 Kbps)
38
38
20
50
31
39
ilbc_mode_30 (13.33 Kbps)
50
50
30
33.3
24
29
gsm (13 Kbps)
33
33
20
50
30
37
gsm (12 Kbps)
32
32
20
50
29
37
G.Clear (64 Kbps)
80
160
20
50
80
88
GSM AMR
--
--
--
--
15
15
ISAC (32 Kbps)
--
--
--
--
37
37
Aacld (mpeg4)
--
--
--
--
Derived from the SDP bandwidth attribute (TIAS)
Derived from the SDP bandwidth attribute (TIAS)
Table 2
Fax Bandwidth Table
T.38 Fax Bit Rate
Redundancy
Maximum Bandwidth in Kbps
2400
None
8
2400
Redundancy
17
9600 (default)
None
16
9600 (default)
Redundancy
46
14400
None
20
14400
Redundancy
65
33600
None
40
33600
Redundancy
142
How to Configure Bandwidth-Based Call Admission Control
Configuring Bandwidth-Based Call Admission Control at the Interface Level
You can configure the Bandwidth-Based Call Admission Control feature at the interface level to reject SIP calls when the bandwidth required for the call exceeds the aggregate bandwidth threshold.
You can configure the Bandwidth-Based Call Admission Control feature for the following interfaces:
ATM
Ethernet (Fast Ethernet, Gigabit Ethernet)
Loopback
Serial
Note
Cisco recommends that you configure a bind media to associate a specific interface for SIP calls. Otherwise, the interface used for the calls will be determined based on the best local address that can access the remote media source address (for early offer calls) or the remote signaling source address (for delayed offer calls). When you use a Loopback interface to configure CAC, you must configure an additional bind-to-bind media with the Loopback interface at the global level or the dial peer level. Configure the
bind media source-interface loopbacknumber command in service SIP configuration mode to configure a bind media.
Configures the Bandwidth-Based Call Admission Control feature at the interface level to reject SIP calls when the bandwidth required for the calls exceed the aggregate bandwidth threshold.
You can configure the
call threshold interfacetype numberlowlow-thresholdhighhigh-threshold [midcall-exceed] command to apply call admission control to reject SIP calls once the accounted bandwidth reaches the
high-threshold value and continues to be above the
low-threshold value.
You can configure the
call threshold interfacetype numberint-bandwidthclass-mapname [l2-overheadpercentage] [midcall-exceed] command to use the bandwidth value provisioned in the QoS policy under the interface for VoIP media traffic for CAC. See the Modular Quality of Service Command-Line Interface Overview document at
http://www.cisco.com/en/US/docs/ios/12_2/qos/configuration/guide/qcfmdcli.html for information on the usage of the QoS policy with Call Admission Control.
SIP calls are rejected when the calculated aggregate bandwidth of VoIP media traffic on the specified interface exceeds the configured bandwidth threshold.
Step 4
end
Example:
Device(config)# end
Exits global configuration mode and enters privileged EXEC mode.
Configuring Bandwidth-Based Call Admission Control at the Dial Peer Level
You can configure the Bandwidth-Based Call Admission Control feature at the dial peer level to reject SIP calls when the bandwidth required for the calls exceeds the aggregate bandwidth threshold.
SUMMARY STEPS
1.enable
2.configure terminal
3.dial-peer voicetagvoip
4.session protocol sipv2
5.max-bandwidthbandwidth-value[midcall-exceed]
6.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
dial-peer voicetagvoip
Example:
Device(config)# dial-peer voice 44 voip
Enters dial peer voice configuration mode.
Step 4
session protocol sipv2
Example:
Device(config-dial-peer)# session protocol sipv2
Configures the Bandwidth-Based Call Admission Control feature for SIP dial peers only.
Configures the Bandwidth-Based Call Admission Control feature at the dial peer level to reject SIP calls when the bandwidth required for the calls exceed the aggregate bandwidth threshold.
Configuring the
midcall-exceed keyword allows exceeding the bandwidth threshold during mid-call media renegotiation. Media renegotiation exceeding the bandwidth threshold is rejected by default.
Step 6
end
Example:
Device(config-dial-peer)# end
Exits dial peer configuration mode and enters privileged EXEC mode.
Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping
Mapping of the call rejection cause code to a specific SIP error response code is known as error response code mapping. The cause code for the call rejected because of the bandwidth-based CAC can be mapped to a SIP error response code between 400 to 600. The default SIP error response code is 488.
You can configure SIP error response codes for calls rejected by the Bandwidth-Based Call Admission Control feature at the global level, dial peer level, or both.
Configures bandwidth-based CAC SIP error response code mapping at the dial peer level.
Step 5
end
Example:
Device(config-dial-peer)# end
Exits dial peer configuration mode and enters privileged EXEC mode.
Verifying Bandwidth-Based Call Admission Control
Perform this task to verify the configuration for the Bandwidth-Based Call Admission Control feature on Cisco UBE. The
show commands need not be entered in any specific order.
SUMMARY STEPS
1.enable
2.show call threshold config
3.show call threshold status
4.show call threshold stats
5.show dial-peer voice
DETAILED STEPS
Step 1
enable
Example:
Device>enable
Enables privileged EXEC mode.
Step 2
show call threshold config
Example:
Device# show call threshold config
Some resource polling interval:
CPU_AVG interval: 60
Memory interval: 5
IF Type Value Low High Enable
----- ---- ----- ---- ---- ------
GigabitEthernet0/0 int-bandwidth 0 100 400 N/A
Displays the current call threshold configuration at the interface level for all resources.
Step 3
show call threshold status
Example:
Device# show call threshold status
Status IF Type Value Low High Enable
------ --- ------ ---- ---- ---- -----
Avail GigabitEthernet0/0 int-bandwidth 0 100 400 N/A
Displays the availability status of resources that are configured when the Bandwidth-Based Call Admission Control feature is enabled at an interface level.
Step 4
show call threshold stats
Example:
Device# show call threshold stats
Total resource check: 2
successful: 1
failed: 1
1: ------------------------
Failed resources: int-bandwidth,
related interface: GigabitEthernet0/0; related option:N/A
Recorded time: 04:49:39 UTC Wed Dec 8 2010
2: ------------------------
Successful
All resources are available for this check.
Recorded time: 04:29:39 UTC Wed Dec 8 2010
Displays the statistics of resources that are configured when the Bandwidth-Based Call Admission Control feature is enabled at an interface level.
Example: Configuring Bandwidth-Based Call Admission Control at the Interface Level
The following example shows how to configure Cisco UBE to reject new SIP calls if the accounted VoIP media bandwidth on Gigabit Ethernet interface 0/0 exceeds 400 Kbps of bandwidth and continues to have a bandwidth above 100 Kbps:
The following example shows how to configure Cisco UBE to reject new SIP calls if the VoIP media bandwidth on Gigabit Ethernet interface 0/0 exceeds the configured bandwidth for priority traffic in the "voip_traffic" class:
Layer 2 overhead of 10 percent in the
call threshold command indicates that the IP bandwidth, excluding Layer 2, is 90 percent of the configured priority bandwidth.
Example: Configuring Bandwidth-Based Call Admission Control at the Dial Peer Level
The following example shows how to configure Cisco UBE to reject calls once the accounted aggregate bandwidth of active calls exceeds 400 Kbps for a SIP dial peer:
Feature Information for Bandwidth-Based Call Admission Control
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 3
Feature Information for Bandwidth-Based Call Admission Control
Feature Name
Releases
Feature Information
Bandwidth-Based Call Admission Control
15.2(2)T
The Bandwidth-Based Call Admission Control feature provides the functionality to reject SIP calls when the bandwidth accounted by the SIP signaling layer exceeds the aggregate bandwidth threshold for VoIP media traffic--voice, video, and fax. This functionality helps prevent QoS degradation of VoIP media traffic for existing calls when the bandwidth allocated for VoIP traffic is fully utilized.
The following commands were introduced or modified:
call threshold interface,
error-code-override,
max-bandwidth,
show call threshold,
voice-class sip
Bandwidth-Based Call Admission Control
Cisco IOS XE Release 3.7S
The Bandwidth-Based Call Admission Control feature provides the functionality to reject SIP calls when the bandwidth accounted by the SIP signaling layer exceeds the aggregate bandwidth threshold for VoIP media traffic--voice, video, and fax. This functionality helps prevent QoS degradation of VoIP media traffic for existing calls when the bandwidth allocated for VoIP traffic is fully utilized.
The following commands were introduced or modified:
call threshold interface,
error-code-override,
max-bandwidth,
show call threshold,
voice-class sip
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL:
www.cisco.com/go/trademarks. Third-party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (1110R)
Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.