- Cisco Unified Border Element Enterprise Protocol-Independent Features and Setup
- SIP-to-SIP Extended Feature Functionality for Session Border Controllers
- Bandwidth-Based Call Admission Control
- Interworking Between RSVP Capable and RSVP Incapable Networks
- Cisco Resource Reservation Protocol Agent
- SIP INFO Method for DTMF Tone Generation
- DTMF Events through SIP Signaling
- Call Progress Analysis Over IP-to-IP Media Session
- Codec Preference Lists
- AAC-LD MP4A-LATM Codec Support on Cisco UBE
- Multicast Music-on-Hold Support on Cisco UBE
- Network-Based Recording
- Video Recording - Additional Configurations
- TDoS Attack Mitigation
- Cisco Unified Communications Gateway Services--Extended Media Forking
- Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls
- iLBC Support for SIP and H.323
- DSP-Based Functionality on the Cisco UBE Enterprise Including Transcoding and Transrating
- Acoustic Shock Protection
- Noise Reduction
- SIP Ability to Send a SIP Registration Message on a Border Element
- SIP Profiles
- Session Refresh with Reinvites
- SIP Stack Portability
- VoIP for IPv6
- Interworking of Secure RTP calls for SIP and H.323
- Cisco UBE Support for SRTP-RTP Internetworking
- Support for SRTP Termination
- WebEx Telepresence Media Support Over Single SIP Session
- SIP SRTP Fallback to Nonsecure RTP
- Support for Software Media Termination Point
- Cisco Unified Communication Trusted Firewall Control
- Cisco Unified Communication Trusted Firewall Control-Version II
- Finding Feature Information
- Domain-Based Routing Support on the Cisco UBE
- URI-Based Dialing Enhancements
- Additional References
- Glossary
- Finding Feature Information
- Restrictions for Bandwidth-Based Call Admission Control
- Information About Bandwidth-Based Call Admission Control
- How to Configure Bandwidth-Based Call Admission Control
- Configuring Bandwidth-Based Call Admission Control at the Interface Level
- Configuring Bandwidth-Based Call Admission Control at the Dial Peer Level
- Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping
- Verifying Bandwidth-Based Call Admission Control
- Troubleshooting Tips
- Configuration Examples for Bandwidth-Based Call Admission Control
- Example: Configuring Bandwidth-Based Call Admission Control at the Interface Level
- Example: Configuring Bandwidth-Based Call Admission Control at the Dial Peer Level
- Example: Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Global Level
- Example: Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Dial Peer Level
- Feature Information for Bandwidth-Based Call Admission Control
Bandwidth-Based Call Admission Control
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. |
- Finding Feature Information
- Restrictions for Bandwidth-Based Call Admission Control
- Information About Bandwidth-Based Call Admission Control
- How to Configure Bandwidth-Based Call Admission Control
- Configuration Examples for Bandwidth-Based Call Admission Control
- Feature Information for Bandwidth-Based Call Admission Control
Finding Feature Information
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.
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.
- Non-SIP call legs.
- SIP-to-H32X call flows (SIP-H320, H320-SIP, SIP-H324, H324-SIP).
- Subinterfaces for bandwidth-based CAC on an interface.
Information About Bandwidth-Based Call Admission Control
Maximum Bandwidth Calculation
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.
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) |
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:
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 loopback number command in service SIP configuration mode to configure a bind media. |
1.
enable
2.
configure terminal
3.
call threshold interface
type number
int-bandwidth {class-map
name [l2-overhead
percentage] |
low
low-threshold
high
high-threshold} [midcall-exceed]
4.
end
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Device> enable |
Enables privileged EXEC mode. |
Step 2 | configure terminal
Example: Device# configure terminal |
Enters global configuration mode. |
Step 3 | call threshold interface
type number
int-bandwidth {class-map
name [l2-overhead
percentage] |
low
low-threshold
high
high-threshold} [midcall-exceed]
Example: Device(config)# call threshold interface GigabitEthernet 0/0 int-bandwidth low 1000 high 20000 midcall-exceed or Device(config)# call threshold interface GigabitEthernet 0/0 int-bandwidth class-map voip-traffic l2-overhead 20 midcall-exceed |
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.
|
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.
1.
enable
2.
configure terminal
3.
dial-peer voice
tag
voip
4.
session protocol sipv2
5.
max-bandwidth
bandwidth-value
[midcall-exceed]
6.
end
DETAILED STEPS
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.
- Configuring Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Global Level
- Configuring Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Dial Peer Level
Configuring Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Global Level
1.
enable
2.
configure terminal
3.
voice service voip
4.
sip
5.
error-code-override cac-bandwidth failure
sip-status-code-number
6.
end
DETAILED STEPS
Configuring Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Dial Peer Level
1.
enable
2.
configure terminal
3.
dial-peer voice
tag {pots |
voatm |
vofr |
voip}
4. voice-class sip error-code-override cac-bandwidth failure {sip-status-code-number | system}
5.
end
DETAILED STEPS
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.
1.
enable
2.
show call threshold config
3.
show call threshold status
4.
show call threshold stats
5.
show dial-peer voice
DETAILED STEPS
Troubleshooting Tips
Configuration Examples for Bandwidth-Based Call Admission Control
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:
Device> enable Device# configure terminal Device(config)# call threshold interface GigabitEthernet 0/0 int-bandwidth low 100 high 400
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:
Device>enable Device# configure terminal Device(config)# class-map match-all voip-traffic Device(config-cmap)# policy-map voip-policy Device(config-pmap)# class voip-traffic Device(config-pmap-c)# priority 440 Device(config-pmap-c)# end Device# enaconfigure terminalble Device(config)# call threshold interface GigabitEthernet 0/0 int-bandwidth class-map voip-traffic l2-overhead 10
Note | 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:
Device> enable Device# configure terminal Device(config)# dial-peer voice 2000 voip Device(config)# session protocol sipv2 Device(config-dial-peer)# max-bandwidth 400
Example: Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Global Level
The following example shows how to configure Cisco UBE for bandwidth-based CAC SIP error response code mapping at the global level:
Device> enable Device# configure terminal Device(config)# voice service voip Device(conf-voi-serv)# sip Device(conf-serv-sip)# error-code-override cac-bandwidth 500
Example: Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Dial Peer Level
The following example shows how to configure Cisco UBE for bandwidth-based CAC SIP error response code mapping at the dial peer level:
Device> enable Device# configure terminal Device(config)# dial-peer voice 88 voip Device(config-dial-peer)# voice-class sip error-code-override cac-bandwidth failure 500
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.
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 |