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Cisco Unified Border Element

Cisco Unified Border Element (CUBE) Management and Manageability Specification

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Table of Contents

1 Product/Feature......................................................................................................................... 3

1.1 Overview/Description............................................................................................................................................... 3

2 Embedded Management............................................................................................................ 6

2.1 CLI—Provisioning....................................................................................................................................................... 6

2.1.1 Global CUBE CLI............................................................................................................................................... 6

2.1.2 SIP CLI................................................................................................................................................................. 7

2.1.3 H.323 CLI............................................................................................................................................................. 8

2.1.4 Dial-Peer CLI...................................................................................................................................................... 8

2.1.5 Security Features CLI........................................................................................................................................ 9

2.2 CLI—Status.................................................................................................................................................................. 9

2.2.1 SIP Trunk Status................................................................................................................................................. 9

2.2.2 Call Admission Control................................................................................................................................... 10

2.3 Protocol Monitoring................................................................................................................................................. 11

2.3.1 SIP Resource Availability............................................................................................................................... 11

2.4 SNMP Monitoring...................................................................................................................................................... 12

2.4.1 Router and Interface Health........................................................................................................................... 12

2.4.2 SIP Trunk Status............................................................................................................................................... 13

2.4.3 Call Traffic Statistics........................................................................................................................................ 13

2.4.3.1 Real-Time Trunk Utilization............................................................................................................................ 15

2.4.3.2 Historical Trunk Utilization............................................................................................................................. 17

2.4.3.3 Call Arrival Rate.............................................................................................................................................. 17

2.4.3.4 Call Success/Failure Statistics......................................................................................................................... 17

2.4.3.5 Transcoding Session Capacity and DSP Utilization........................................................................................ 20

2.4.3.6 MTP Session Capacity and DSP Utilization................................................................................................... 21

2.4.4 Licensing and Call Admission Control........................................................................................................ 22

2.4.5 Voice Quality MIBs........................................................................................................................................... 22

2.5 SNMP Traps.................................................................................................................................................................. 23

2.6 Syslog Messages......................................................................................................................................................... 23

2.7 Embedded Event Manager (EEM)............................................................................................................................ 25

2.7.1 SIP Trunk Status............................................................................................................................................... 25

2.8 IP SLA.............................................................................................................................................................................. 27

2.9 NetFlow.......................................................................................................................................................................... 27

3 Supported Management Applications....................................................................................... 29

4 Management Recommendations.............................................................................................. 31

4.1 Provisioning Recommendations............................................................................................................................. 31

4.1.1 Command Line (CLI)...................................................................................................................................... 31

4.1.2 Graphical (GIU)................................................................................................................................................ 31

4.2 SIP Trunk Security Recommendations.................................................................................................................. 33

4.2.1 Service Provider (SP) SIP Trunk Security................................................................................................... 34

4.2.2 Toll Fraud Security........................................................................................................................................... 36

4.2.3 New Security Operation in Cisco IOS 15.1.2T............................................................................................ 36

4.3 Monitoring Recommendations................................................................................................................................. 37

4.4 Troubleshooting Recommendations...................................................................................................................... 38

4.4.1 General.............................................................................................................................................................. 38

4.4.2 High-Traffic-Volume Troubleshooting (PCD)............................................................................................ 40

4.4.3 SIP Ladder Diagrams..................................................................................................................................... 41

5 Glossary.................................................................................................................................. 43

6 References............................................................................................................................... 45

List of Tables

Table 1. Operations Phase and Management Capabilities........................................................................................... 5

Table 2. CISCO-PROCESS-MIB........................................................................................................................................ 13

Table 3. CISCO-MEMORY-POOL-MIB.............................................................................................................................. 13

Table 4. IF-MIB...................................................................................................................................................................... 13

Table 5. CISCO IOS MIBs that Contain Active Call Information.................................................................................. 14

Table 6. CISCO IOS MIBs that Contain Historical Call Information............................................................................ 14

Table 7. Real-Time Trunk Utilization: DIAL-CONTROL-MIB........................................................................................ 15

Table 8. Real-Time Trunk Utilization: CISCO-VOICE-DIAL-CONTROL-MIB............................................................. 15

Table 9. CISCO-VOICE-DIAL-CONTROL-MIB cvCallVolume Information............................................................... 16

Table 10. Historical Trunk Utilization MIB Information.................................................................................................... 17

Table 11. Call Arrival Rate MIB Information...................................................................................................................... 17

Table 12. Call Arrival Rate: CISCO-VOICE-DIAL-CONTROL-MIB................................................................................. 17

Table 13. Call Success/Failure MIB Information.............................................................................................................. 17

Table 14. CISCO-SIP-UA-MIB MIB Fields for 4xx, 5xx and 6xx SIP Responses........................................................ 18

Table 15. Transcoding Session Capacity and DSP Utilization MIB Information........................................................ 20

Table 16. Transcoding Session Capacity and DSP Utilization MIB Information........................................................ 21

Table 17. Voice Quality: CISCO-VOICE-DIAL-CONTROL-MIB...................................................................................... 23

Table 18. Voice Quality: RTTMON MIBs............................................................................................................................. 23

Table 19. Syslog Error Message Severity Levels............................................................................................................. 24

Table 20. Cisco IOS IP SLAs Operations and Applications............................................................................................ 27

Table 21. Supported Management Applications.............................................................................................................. 29

Table 22. Key “show” Commands on Cisco UBE............................................................................................................ 37

Table 23. Key “debug” Commands on Cisco UBE.......................................................................................................... 39


1 Product/Feature

This Cisco Unified Border Element (Cisco UBE) Manageability Document contains information about the Simple Network Management Protocol (SNMP) MIBs, critical system log (syslog) messages and general Cisco IOS commands for monitoring and troubleshooting a Cisco UBE deployment. Cisco UBE is a Cisco IOS feature set supported on the Integrated Services Router (ISR) and Aggregation Services Routers (ASR) series platforms.

Cisco UBE is an integrated Cisco IOS enterprise session border controller (SBC) feature set facilitating simple and cost-effective connectivity between independent unified communications, voice over IP (VoIP), and video networks. Typical connectivity deployments where Cisco UBE is used include:

Connect Cisco Unified Communication Manager (CUCM) enterprises to service provider SIP trunks

Connect 3rd party IP-PBX enterprises to service provider SIP trunks

Connect H.323 and SIP voice and video applications within the enterprise

Connect H.323 video over the Internet into the enterprise

Connect business-to-business TelePresence sessions between enterprises

Session border controllers (SBCs), such as Cisco UBE, offer unified communications network interoperability features such as:

Session Management: Offers real-time session management at the network border, such as call admission control, dial-plan interpretation and routing, SLA monitoring, QoS policy marking, etc.

Interworking: Offers feature to interconnect networks with different protocols or capabilities, such as H.323-SIP interworking, SIP normalization, DTMF type conversion, payload type conversion, IPv4-IPv6 interworking, transcoding and transrating, etc.

Demarcation: Allows a single point of troubleshooting for SIP trunks and voice quality issues. Offers features such topology hiding, statistics and billing (call detail records, or CDR) at the border of the network.

Security: Offers a security enforcement point at the network border through features such as SIP registration, SIP port protection, hostname validation, authentication and encryption features, etc.

1.1 Overview/Description

Four aspects of Cisco UBE Manageability are addressed in this document:

1. Image, Configuration and License Management: General Cisco IOS router tools and methods are used for this. Cisco UBE licensing is in effect, but is only enforced for Gatekeeper configurations (as of 12.4.20T) and not yet for Cisco UBE configurations. When deploying any Unified Communications feature, including Cisco UBE, on and ISR G2 platform, the UC Technology Package is required. The licensing for this package is enforced.

See Cisco IOS Software Activation for more details.

2. Provisioning: General Cisco IOS router provisioning using the command line interface (CLI) is supported.

Support by management provisioning tools such as Cisco Configuration Professional (CCP). CCP 2.3 introduces support for Cisco UBE provisioning.

Cisco UBE provisioning includes the following elements:

General router attributes

- Routing protocols, router interfaces, access lists, DNS connectivity, NTP (clock settings), QoS policies, SNMP connectivity, AAA/RADIUS connectivity, security features, etc.


Global Cisco UBE attributes

- Turn on Cisco UBE as a router functions and specify the protocols that should be handled

- DSP hardware configuration and attributes (if present)

SIP provisioning

- Global SIP parameters and attributes

- SIP header manipulation

- Dial-peers for SIP call sources and destinations

- SIP User Agent parameters and attributes

H.323 provisioning

- Global H.323 parameters and attributes

- Dial-peers for H.323 call sources and destinations

Dial-Plan provisioning

- Dial-peers, translation rules and digit manipulation features for interpreting the dial plan and routing calls as desired

3. Monitoring: General Cisco IOS router monitoring using CLI, syslog and SNMP are supported.

Cisco UBE supports most of the general Cisco IOS unified communications SNMP MIBs as well as several OIDs (object identifiers) developed specifically for Cisco UBE use cases.

Cisco UBE monitoring includes the following elements:

Router Inventory and Health: CPU, memory, flash, modules, software image and release, etc.

Interface Health: General IOS router interfaces, status and packet traffic statistics.

SIP Trunk Status: Up or Down status of a SIP trunk to a service provider or application

Call Traffic Statistics (Calls, Sessions, Capacity Planning, Errors):

- Trunk utilization and H.323/SIP Session Capacity

- Call arrival rate

- Call success/failure statistics

- SIP retries statistics

- Transcoding Session Capacity and DSP Utilization

- Media Termination Point (MTP) Session Capacity

Licensing and Call Admission Control

Resource Availability: Statistics and feedback to upstream call agents and load balancers

Voice Quality: Statistics on packet loss, delay and jitter that can be calculated into metrics such as ICPIF, MOS and R-factor scores

Billing: CDR, call patterns, toll fraud monitoring

4. Troubleshooting: General Cisco IOS router troubleshooting using CLI show and debug commands, as well as packet capture methods, are supported.

Cisco UBE supports most of the general Cisco IOS unified communications show and debug commands as well as several commands, and extensions to existing commands, developed specifically for Cisco UBE use cases, such as Per Call Debugging (PCD).

Table 1 provides an overview of Cisco UBE management capabilities that can be used during different operations phases.

Table 1. Operations Phase and Management Capabilities

Operations Phase

Management Capability

Staging/Configuration

Cisco IOS CLI
Cisco Configuration Engine (CCE)
Configuration examples at www.cisco.com/go/interoperability > Cisco Unified Border Element/SIP Trunking Solutions
Cisco Configuration Professional 2.3 or later

Installation/Provisioning

Cisco IOS CLI
Configuration examples at www.cisco.com/go/interoperability > Cisco Unified Border Element/SIP Trunking Solutions

Change Management/Archiving

Cisco IOS CLI

Fault Monitoring/Management

Cisco IOS CLI
Any SNMP-based management system

Performance Monitoring/Management

Cisco IOS CLI (show and debug commands)
Cisco UBE CDR
Any SNMP-based management system

Troubleshooting

Cisco IOS CLI (show and debug commands)
Cisco IOS syslog
Wireshark (open source application)


2 Embedded Management

Key embedded management capabilities of the Cisco IOS router where Cisco UBE is deployed are covered in this section. This includes:

CLI

SNMP

Syslog

IP SLA

EEM

NetFlow

2.1 CLI—Provisioning

This section summarizes the key or common Cisco UBE CLI used to provision basic system functionality. Most specialized Cisco UBE features and deployments have additional CLI to turn on specific features. General Cisco IOS router configuration is assumed known and is not covered here.

Additional in-depth Cisco UBE configuration resources include:

Cisco UBE IOS configuration is fully documented at: www.cisco.com/go/cube > Configure > Configuration Guides > Cisco Unified Border Element Configuration Guide.

Cisco UBE IOS configuration is fully documented at: www.cisco.com/go/cube > Configure > Configuration Guides > Cisco Unified Border Element Configuration Guide.

Cisco UBE CLI commands are documented as part of the general Cisco IOS command reference documentation on Cisco.com

Cisco UBE configuration examples are given at www.cisco.com/go/cube > Configure > Configuration Guides > Configuration Examples and TechNotes

Cisco UBE interoperability configuration guides with service provider SIP trunks and 3rd party IP-PBXs are given at www.cisco.com/go/interoperability > Cisco Unified Border Element (CUBE)/SIP Trunking Solutions

Note: Please refer to the general Cisco IOS command references on Cisco.com for a full explanation of command options and syntax, only abbreviated examples are given in the following sections.

2.1.1 Global CUBE CLI

Several attributes of CUBE are configured at the global level of the router. This includes generic capabilities, as well as global SIP and H.323 capabilities.

Basic routing, connectivity and access lists are required as pre-requisite router configuration for CUBE. Additional generic router capabilities such as DHCP, QoS or firewall are optional.

Enable CUBE (all platforms):

Cisco UBE is being deployed on a Cisco IOS router when one of the following commands is present:

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip

Enable CUBE (required on ISR G2):

Cisco UBE is turned on for an ISR G2 platform with the following command:

voice service voip
mode border-element

Fax:

Fax configuration on Cisco UBE uses the same CLI as fax control commands for Cisco IOS PSTN gateways. This includes both global and dial-peer commands.

More information can be found in the Cisco IOS Fax, Modem, and Text Support over IP Configuration Guide.

Call Admission Control (CAC):

Cisco UBE supports global or interface-level CAC based on call count, CPU or memory use by using the following CLI:

call threshold global
call threshold interface
call treatment on

Cisco UBE can detect (and alter behavior) spikes in call arrival rate (useful for SIP DOS protection) by using the following CLI:

call spike

Cisco UBE supports destination-specific limits on call counts by using the following CLI:

dial-peer voice x voip
max-connection

Transcoding:

Cisco UBE can use DSPs to provide transcoding and transrating services. This configuration is covered in detail in the Cisco UBE configuration examples given at www.cisco.com/go/cube > Configure > Configuration Guides > Configuration Examples and TechNotes > Unified Border Element Transcoding Configuration Example.

2.1.2 SIP CLI

Several SIP attributes of Cisco UBE are configured at the global level of the router and applies to all SIP communications. Many Cisco UBE features have both global and dial-peer CLI so that they can easily be turned on globally if the function is needed on all calls or turned on/off per call destination if more granular or policy control of call handling is needed.

voice service voip
sip
address-hiding
bind control
bind media
session transport
rel1xx
header-passing
midcall-signaling passthrough
sip-ua
authentication
credentials
registrar
sip-server
retry invite
retry register
timers connect

2.1.3 H.323 CLI

Several H.323 attributes of Cisco UBE are configured at the global level of the router and applies to all H.323 communications. Many Cisco UBE features have both global and dial-peer CLI so that they can easily be turned on globally if the function is needed on all calls or turned on/off per call destination if more granular or policy control of call handling is needed.

H.323-H.323:

voice service voip
no supplementary-service h450.2 ! Disable call transfer with H.450
no supplementary-service h450.3 ! Disable call forward with H.450
no supplementary-service h450.7
no supplementary-service h450.12 ! Hidden CLI
supplementary-service media-renegotiate ! Enables media renegotiation in case
! of Refer to ECS
supplementary-service ringback h225-info ! Enables Ringback
h323
emptycapability ! Enables supplementary services using ECS
h245 passthru tcsnonstd-passthru ! Interop with CUCM to pass-through
! non-standard parameters
h225 connect-passthrough ! Required for H323-H323 calls with CUCM

Additional commands for H.323-H.323:

voice service voip
address-hiding
allow-connections h323 to h323

Additional commands for H.323-SIP:

voice service voip
address-hiding
allow-connections h323 to sip
allow-connections sip to h323

2.1.4 Dial-Peer CLI

Cisco UBE dial-plan interpretation and call routing is implemented using VoIP dial-peers and the configuration is very similar to that of a Cisco IOS PSTN gateway. Translation rules and digit manipulation features are supported on both deployments.

Please refer to the Cisco.com Cisco IOS Dial Peer Configuration on Voice Gateway Routers configuration guide for details of available dial plan implementation commands.

H.323 is the default protocol for a dial-peer in Cisco IOS. To enable SIP as the protocol, use the following command:

dial-peer voice x voip
session protocol sipv2

As of Cisco UBE 8.5 (IOS 15.1.2T), the source IP address used in SIP messaging can be controlled per dial-peer by using the following CLI:

dial-peer voice x voip
session protocol sipv2
voice-class sip bind control
voice-class sip bind media

2.1.5 Security Features CLI

Some Cisco UBE-specific security features, such as topology hiding and protocol stack protection (detecting malformed and rogue packets) are enabled and active by default. Many other features are not enabled by default and require CLI to mitigate against targeted attacks or security breaches. Like any other network and router device, Cisco UBE should be locked down against security attacks. Please see the later section on “Security Recommendations” for guidelines on feature to turn on.

2.2 CLI—Status

2.2.1 SIP Trunk Status

SIP trunk status is an important element of CUBE monitoring. SIP Trunk status can be monitored by configuring an out-of-dialog (OOD) SIP Options PING as a keepalive mechanism on the dial-peer(s) pointing towards the SIP Trunk, using the CLI example below.

dial-peer voice 100 voip
destination-pattern .T
voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6
session protocol sipv2
session target ipv4:x.x.x.x

When calls to the SIP trunk are successful, the dial-peer is in “active” state. If SIP PING timeouts occur, the dial-peer changes to “busyout” status. Calls to the dial-peer during “busyout” will be rejected immediately to the originator for call rerouting.

CUBE 1.3 (Cisco IOS 15.0.1M) returns an unconfigurable SIP “404 Not Found” error code

CUBE 1.4 (15.1.1T) or later allows a configurable SIP error code in the 400-699 range. The default is “503 Service Unavailable”

Dial-peer state changes are as follows:

Dial-peer is marked as “active” when a valid response to an Options PING is received

Dial-peer is marked as “busyout” when no response to an Options PING is received

Dial-peer status changes from “active” to “busyout” when:


- A “503 Service Unavailable” response is received

- No response is received, i.e. request timeout (configurable number of retries)

- A “505 Version not supported” response is received

Dial-peer status changes from “busyout” to “active” after a configurable number of consecutive positive responses (i.e. anything except 503, 505 and t/o)

On router reboot, all dial-peers start in the “active” state

The CLI to configure a SIP OOD Options PING is:

voice service voip
sip
error-code-override options-keepalive failure 500
dial-peer voice 10 voip
voice-class sip error-code-override options-keepalive failure 500

The dial-peer status based on the SIP OOD Options PING can be displayed with the following “show” commands:

router# show dial-peer voice summary
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT KEEPALIVE
1 voip up up 1000 0 syst ipv4:x.x.x.10 active
2 voip up up 2000 0 syst ipv4:x.x.x.11 busyout
3 voip up up 3000 0 syst ipv4:x.x.x.12

router# show dial-peer voice | include options
voice class sip options-keepalive up-interval 100 down-interval 50 retry 6
voice class sip options-keepalive dial-peer action = active,
voice class sip options-keepalive up-interval 100 down-interval 50 retry 6
voice class sip options-keepalive dial-peer action = busyout,

In CUBE releases older than CUBE 1.3 (15.0.1M), or in addition to the layer 7 SIP monitoring described above, a layer 3 connectivity monitoring can be done using an ICMP ping. The following CLI can be used to enable this feature:

dial-peer voice 10 voip
destination-pattern .T
monitor probe icmp-ping x.x.x.x
session protocol sipv2
session target ipv4:x.x.x.x

2.2.2 Call Admission Control

Call rejections due to CAC threshold being met or exceeded can be seen by using the following show commands:

show call spike status
show call threshold status
show call admission statistics
show call treatment stats

2.3 Protocol Monitoring

Some statistics or traffic information are embedded within the SIP or H.323 call control protocol. These are covered in this section.

Statistics and feedback to upstream call agents and load balancers are provided by Cisco UBE so that these network elements can adjust their call routing and load balancing algorithms based on the load experienced by the session border controller (Cisco UBE). One such method is the Resource Availability Indicator (RAI), available in both H.323 and SIP.

2.3.1 SIP Resource Availability

RAI for SIP is implemented as of CUBE 8.5 (Cisco IOS release 15.1.2T). Cisco UBE resources that can be monitored using this method include:

System

CPU

Memory

DSP

The method uses an Out-of-Dialog SIP OPTIONS PING message Cisco UBE to the upstream call agent or load balancer. The SIP RAI notification can be initiated by any of these methods:

Unsolicited (based on static Cisco UBE configuration)

- Periodically based on a timer configuration

- Notification when a threshold value (low/high water mark configuration) is crossed for a given resource

Solicited (polled, or query/response)

- An SIP application can request RAI information

The following is a sample configuration for unsolicited (based on configuration) periodic RAI reporting.

voice class resource-group 1
resource cpu 1-min-avg
resource dsp
resource mem total-mem
periodic-report interval 30
!
sip-ua
rai target ipv4:9.13.40.83 resource-group 1

The following is a sample configuration for unsolicited (based on configuration) threshold-based RAI reporting.

voice class resource-group 2
resource cpu 1-min-avg threshold high 50 low 30
resource dsp threshold high 50 low 30
resource mem total-mem threshold high 50 low 30
!
sip-ua
rai target ipv4:9.13.40.83 resource-group 2

A SIP application can also poll for RAI status. In this case it sends an SIP OPTIONS PING to Cisco UBE which responds with the resource information on a 200-OK message. For this, the following configuration is needed:

An example of the configuration of the upstream entity to report the RAI to is as follows:

sip-ua
rai target ipv4:x.x.x.x resource-group x

2.4 SNMP Monitoring

Simple Network Management Protocol (SNMP) is based on the manager/agent model consisting of an SNMP manager, an SNMP agent, a database of management information, managed SNMP devices and the network protocol. The SNMP manager provides the interface between the human network manager and the management system. The SNMP agent provides the interface between the manager and the physical device being managed.

An SNMP-managed network consists of the following:

Managed Device: A network node that contains an SNMP agent that resides on a managed network. Managed devices collect and store management information and use SNMP to make this information available to the NMS. Managed devices, sometimes called network elements, can include routers and access servers, switches and bridges, hubs, computer hosts, and printers.

Agent: A network management software module that resides in a managed device. An agent has local knowledge of management information and translates that information into a form compatible with SNMP.

NMS: Executes applications that monitor and control managed devices. NMSs provide most of the processing and memory resources required for network management. Every managed network must have one or more NMS.

The SNMP agent exchanges network management information with the SNMP manager software that is running on a network management system (NMS). The agent responds to requests for information and actions from the managed device (in this case the Cisco UBE router). The agent controls access to the agent’s MIB, the collection of objects that can be viewed or changed by the SNMP manager. By polling managed devices, an SNMP manager collects information on network connectivity, activity, and events.

Cisco UBE monitoring via SNMP includes the following capabilities:

Router and Interface Health

Call Traffic Reports

- Trunk utilization and H.323/SIP Session Capacity

- Call arrival rate

- Call success/failure statistics

- SIP retries statistics

- Transcoding Session Capacity and DSP Utilization

- MTP Session Capacity

Licensing and Call Admission Control

Voice Quality: Statistics on packet loss, delay and jitter that can be calculated into metrics such as ICPIF, MOS and R-factor scores

2.4.1 Router and Interface Health

These MIBs/OIDs allow you to monitor the physical chassis, interface connectivity, CPU and memory.

Router Inventory and Health: CPU, memory, flash, modules, software image and release, etc.

Interface Health: General IOS router interfaces, status and packet traffic statistics.

Critical router functions, like routing protocol processing and process packet switching, are handled in memory and share the CPU. Thus, if CPU utilization is very high, it is possible that a routing update cannot be handled or packets are dropped. The CISCO-PROCESS-MIB reports the percentage of the processor in use over a five-minute average.

Table 2. CISCO-PROCESS-MIB

OID

OID#

New/Changed

Platform

Use/Operation

cpmCPUTotal5minRev

1.3.6.1.4.1.9.9.109.1.1.1.1.8

Original IOS

All

Health

Memory use can be monitored using the CISCO-MEMORY-POOL-MIB.

Table 3. CISCO-MEMORY-POOL-MIB

OID

OID#

New/Changed

Platform

Use/Operation

ciscoMemoryPoolEntry

1.3.6.1.4.1.9.9.48.1.1.1

Baseline

All

Health Monitoring

The status of physical interfaces on the router platform can be monitored using the IF-MIB.

Table 4. IF-MIB

OID

OID#

New/Changed

Platform

Use/Operation

IfEntry

1.3.6.1.2.1.2.2.1

Baseline

All

Fault Monitoring

2.4.2 SIP Trunk Status

SIP trunk status is an important element of CUBE monitoring. This status is not currently available via SNMP (only via CLI as covered in the previous section).

2.4.3 Call Traffic Statistics

A key element of CUBE monitoring is call traffic reports, both for the volume of calls over time, or for monitoring of call arrival rates. This is useful for various business purposes, including:

Trunk utilization, both real-time and historical

Capacity planning

Troubleshooting

Highlighting errors occurring in call routing or call handling that may indicate a network outage, dial-plan deficiencies, or perhaps an architectural call flow that is not implemented correctly

Detecting call spikes, caused by both normal (an uptick in traffic due to an advertisement or other business event) and malicious (a SIP DOS attack) traffic patterns

CUBE call traffic reports can be provided by information in several SNMP MIBs, some of which have been available historically in all Cisco IOS releases, and others specifically introduced with CUBE 1.4 to aid in traffic reporting. For best results, using CUBE 1.4 or later is recommended.

Trunk utilization and H.323/SIP session capacity statistics

Call arrival rate statistics

Call success/failure statistics

SIP error and timeout/retry statistics

DSP utilization and transcoding session capacity

MTP utilization and session capacity

Cisco IOS voice/video call SNMP information is generally kept in the set of MIBs given below. These MIBs are used for TDM voice calls as well as VoIP, VoFR and VoATM calls. Some OIDs are only populated for certain types of calls. On the Cisco ISR platforms, these MIBs have been supported for a long time, for Cisco UBE on the Cisco ASR 1000 Series platforms, they are supported as of release 3.1.0.

DIAL-CONTROL-MIB

CISCO-DIAL-CONTROL-MIB

CISCO-VOICE-DIAL-CONTROL-MIB

CISCO-VOICE-COMMON-DIAL-CONTROL-MIB

CISCO-CALL-HISTORY-MIB (this MIB is only populated for ISDN calls on TDM voice gateways and therefore does not apply to Cisco UBE and will not be discussed further here.)

These MIBs generally provide information on:

Currently Active Calls: Real-time statistics of call activity

Call History: Historical statistics after calls have disconnected (similar to CDR)

Table 5. CISCO IOS MIBs that Contain Active Call Information

MIB

OID

Table Name

Number of Entries per Call

DIAL-CONTROL-MIB

1.3.6.1.2.1.10.21

callActiveTable

2

CISCO-VOICE-COMMON-DIAL-CONTROL-MIB

1.3.6.1.4.1.9.10.55

cvCommonDcCallActiveTable

2

CISCO-VOICE-DIAL-CONTROL-MIB

1.3.6.1.4.1.9.9.63

cvCallActiveTable

0 (Used for TDM GW calls only)

CISCO-VOICE-DIAL-CONTROL-MIB

1.3.6.1.4.1.9.9.63

cvVoIPCallActiveTable

2

Table 6. CISCO IOS MIBs that Contain Historical Call Information

MIB

OID

Call History

Number of Entries per Call

DIAL-CONTROL-MIB

1.3.6.1.2.1.10.21

callHistoryTable

Not implemented, do not use

CISCO-DIAL-CONTROL-MIB

1.3.6.1.4.1.9.10.25

cCallHistory

2

CISCO-VOICE-COMMON-DIAL-CONTROL-MIB

1.3.6.1.4.1.9.10.55

cvCommonDcCallHistoryTable

2

CISCO-VOICE-DIAL-CONTROL-MIB

1.3.6.1.4.1.9.9.63

cvCallHistoryTable

0 (Used for TDM GW calls only)

The following “show” commands provide information on active voice, video and fax calls in the system:

sh call active ?
fax show all active calls for fax store & forward
media show all active calls for media
video show all active calls for video
voice show all active calls for voice
!
sh call active video ?
brief show brief version of active video calls
compact show compact version of active video calls
id show only call with specified id
!
sh call active fax ?
brief show brief version of active fax calls
compact show compact version of active fax calls

2.4.3.1 Real-Time Trunk Utilization

Various aspects of real-time Trunk Utilization statistics on currently active calls are available from the MIBs and OIDs covered in this section.

When a callActiveTable (1.3.6.1.2.1.10.21.1.3.1) entry is created for a call, an associated cvCallActiveTable (1.3.6.1.4.1.9.9.63.1.3.1) and cvCommonDcCallActiveTable(1.3.6.1.4.1.9.10.55.1.1.1) entries are created. They are indexed by the callActiveSetupTime (1.3.6.1.2.1.10.21.1.3.1.1.1) and callActiveIndex (1.3.6.1.2.1.10.21.1.3.1.1.2) as defined in DIAL-CONTROL-MIB.

The DIAL-CONTROL-MIB provides:

RFC-2128 information

Monitoring of active calls on a particular dial peer

Packet received/transmitted statistics for active calls

The usefulness of the DIAL-CONTROL-MIB entries (callActiveTransmitPackets, callActiveTransmitBytes, callActiveReceivePackets, callActiveReceiveBytes) is essentially for packet received and transmitted statistics. For most other call parameters, the information provided in the CISCO-VOICE-DIAL-CONTROL-MIB is most useful.

Table 7. Real-Time Trunk Utilization: DIAL-CONTROL-MIB

OID

OID#

New/Changed

Platform

Use/Operation

dialCtlPeerStatsTable

1.3.6.1.2.1.10.21.1.2.2

Baseline

ISR G1, ISR G2, AS5x00, ASR

Provides statistics on overall dial-peer use, indexed by dial-peer number.

callActiveTable

1.3.6.1.2.1.10.21.1.3.1

Baseline

ISR G1, ISR G2, AS5x00, ASR

Provides packet statistics on active calls, indexed by callActiveSetupTime and callActiveIndex.

The CISCO-VOICE-DIAL-CONTROL-MIB provides the number of active calls based on:

Protocol (H.323 or SIP)

Dial-Peer

Interface

Table 8. Real-Time Trunk Utilization: CISCO-VOICE-DIAL-CONTROL-MIB

OID

OID#

New/Changed

Platform

Use/Operation

cvVoIPCallActiveTable

1.3.6.1.4.1.9.9.63.1.3.2

Baseline

ISR G1, ISR G2, AS5x00, ASR

Provides statistics on active VoIP calls, indexed by callActiveSetupTime and callActiveIndex.

cvCallVolume

1.3.6.1.4.1.9.9.63.1.3.8

CUBE 1.4

ISR G1, ISR G2, AS5x00

Total, per-protocol, per-dial-peer and per-interface call active statistics.

Total Trunk Utilization:

A snapshot summary of overall call active statistics on the platform is given by the cvCallVolConnTotalActiveConnections (1.3.6.1.4.1.9.9.63.1.3.8.2) OID.

More detailed information per call (packet statistics, VAD, SRTP, etc.) is given in the cvVoIPCallActiveEntry (1.3.6.1.4.1.9.9.63.1.3.2.1) OID.

Trunk Utilization by Protocol:

A snapshot of call active statistics per protocol is given by the cvCallVolConnEntry (1.3.6.1.4.1.9.9.63.1.3.8.1.1) OID. The cvCallVolConnIndex (1.3.6.1.4.1.9.9.63.1.3.8.1.1.1) defines the protocol type, H.323 = 1, SIP = 2. Therefore:

H.323 (1) call statistics are in the CvCallVolConnActiveConnection.1 (1.3.6.1.4.1.9.9.63.1.3.8.1.1.2.1) OID

SIP(2) call statistics are in the CvCallVolConnActiveConnection.2 (1.3.6.1.4.1.9.9.63.1.3.8.1.1.2.2) OID

Trunk Utilization by Dial-Peer:

A snapshot of call active statistics per dial-peer is given by the cvCallVolPeerEntry (1.3.6.1.4.1.9.9.63.1.3.8.4.1) OID. This table augments the dial-peer configuration table in the cvPeerCfgTable (1.3.6.1.4.1.9.9.63.1.2.1) OID, and uses the dial-peer tag as an index. Therefore:

Incoming call statistics for dial-peer 200 are in the cvCallVolPeerIncomingCalls.200 (1.3.6.1.4.1.9.9.63.1.3.8.4.1.1.200) OID

Outgoing call statistics for dial-peer 200 are in cvCallVolPeerOutgoingCalls.200 (1.3.6.1.4.1.9.9.63.1.3.8.4.1.2.200) OID

Trunk Utilization by Interface:

A snapshot of call active statistics per interface is given by the cvCallVolIfTableEntry (1.3.6.1.4.1.9.9.63.1.3.8.5.1) OID. This table is indexed by the interface number using the ifIndex (1.3.6.1.2.1.2.2.1.1) OID in the IF-MIB. Therefore:

Incoming call statistics for interface 5 are in the cvCallVolMediaIncomingCalls.5 (1.3.6.1.4.1.9.9.63.1.3.8.5.1.1.5) OID

Outgoing call statistics for interface 5 are in cvCallVolMediaOutgoingCalls.5 (1.3.6.1.4.1.9.9.63.1.3.8.5.1.2.5) OID

The cvCallVolume OID in the CISCO-VOICE-DIAL-CONTROL-MIB contains the following call volume information:

Table 9. CISCO-VOICE-DIAL-CONTROL-MIB cvCallVolume Information

OID

OID#

Use/Operation

cvCallVolConnIndex

1.3.6.1.4.1.9.9.63.1.3.8.1.1.1

Index to the cvCallVolConnTable. A value of 1 denotes H.323 calls, and 2 SIP calls.

cvCallVolConnActiveConnection

1.3.6.1.4.1.9.9.63.1.3.8.1.1.2

Number of calls active of the type determined by cvCallVolConnIndex.

cvCallVolConnTotalActiveConnections

1.3.6.1.4.1.9.9.63.1.3.8.2

Total number of active calls on the platform.

cvCallVolPeerIncomingCalls

1.3.6.1.4.1.9.9.63.1.3.8.4.1.1

Number of active incoming calls for a dial-peer.

cvCallVolPeerOutgoingCalls

1.3.6.1.4.1.9.9.63.1.3.8.4.1.2

Number of active outcoming calls for a dial-peer.

cvCallVolMediaIncomingCalls

1.3.6.1.4.1.9.9.63.1.3.8.5.1.1

Number of active incoming calls for an interface.

cvCallVolMediaOutgoingCalls

1.3.6.1.4.1.9.9.63.1.3.8.5.1.2

Number of active outgoing calls for an interface.

2.4.3.2 Historical Trunk Utilization

Historical Trunk Utilization statistics on completed calls are available from the MIBs and OIDs covered in this section. Alternatively, you can also use the Real-time Trunk Utilization statistics in the previous section and store this info to provide your own aggregation and trending information. Up to 1200 call history records are stored in memory in a circular buffer.

Table 10. Historical Trunk Utilization MIB Information

MIB

OID

OID#

New/Changed

Platform

CISCO-DIAL-CONTROL-MIB

cCallHistoryTable

1.3.6.1.4.1.9.10.25.1.4.3

Baseline

ISR G1, ISR G2, AS5x00, ASR

CISCO-VOICE-DIAL-CONTROL-MIB

cvVoIPCallHistoryTable

1.3.6.1.4.1.9.9.63.1.4.2

Baseline

ISR G1, ISR G2, AS5x00, ASR

CISCO-VOICE-COMMON-DIAL-CONTROL-MIB

cvCommonDcCallHistory

1.3.6.1.4.1.9.10.55.1.2.1

Baseline

ISR G1, ISR G2, AS5x00, ASR

2.4.3.3 Call Arrival Rate

Call arrival rate and call spikes can be monitored as of CUBE 1.4 (15.1.1T) or later using the CISCO-VOICE-DIAL-CONTROL-MIB MIB information covered in this section.

Table 11. Call Arrival Rate MIB Information

OID

OID#

New/Changed

Platform

cvCallRateMonitor

1.3.6.1.4.1.9.9.63.1.3.11

CUBE 1.4

ISR G1, ISR G2, AS5x00

By default call rate information is not gathered and the MIB information is empty. To turn on call rate monitoring, use the cvCallRateMonitorEnable (1.3.6.1.4.1.9.9.63.1.3.11.1) OID and set the monitoring period with the cvCallRateMonitorTime (1.3.6.1.4.1.9.9.63.1.3.11.2) OID. There is no facility to turn monitoring on or off via CLI.

The cvCallRateMonitor OID in the CISCO-VOICE-DIAL-CONTROL-MIB contains the following call rate information.

Table 12. Call Arrival Rate: CISCO-VOICE-DIAL-CONTROL-MIB

OID

OID#

Use/Operation

cvCallRateMonitorEnable

1.3.6.1.4.1.9.9.63.1.3.11.1

A value of TRUE starts computation of call rate information. A value of FALSE turns it off.

cvCallRateMonitorTime

1.3.6.1.4.1.9.9.63.1.3.11.2

Value can from 1 to 12—each value denotes a time unit of 5 seconds. That is, a value of 1 means 5 seconds, a value of 2 means 10 seconds, etc.

cvCallRate

1.3.6.1.4.1.9.9.63.1.3.11.3

Number of calls connected during the last monitoring period duration.

cvCallRateHiWaterMark

1.3.6.1.4.1.9.9.63.1.3.11.4

Peak value in any given cvCallRateMonitorTime duration if cvCallRateMonitorEnable is set to TRUE.

2.4.3.4 Call Success/Failure Statistics

Successful and failed call counts can be monitored for trending or troubleshooting purposes using the MIB information covered in this section.

Table 13. Call Success/Failure MIB Information

MIB

OID

OID#

New/Changed

Platform

DIAL-CONTROL-MIB

dialCtlPeerStatsTable

1.3.6.1.2.1.10.21.1.2.2

Baseline

ISR G1, ISR G2, AS5x00, ASR

CISCO-SIP-UA-MIB

cSipStats

1.3.6.1.4.1.9.9.152.1.2

Baseline

ISR G1, ISR G2, AS5x00, ASR

The DIAL-CONTROL-MIB provides information per dial-peer for both H.323 and SIP using the following OIDs:

Success

- dialCtlPeerStatsSuccessCalls (1.3.6.1.2.1.10.21.1.2.2.1.3)

- dialCtlPeerStatsAcceptCalls (1.3.6.1.2.1.10.21.1.2.2.1.5)

Failure

- dialCtlPeerStatsFailCalls (1.3.6.1.2.1.10.21.1.2.2.1.4)

- dialCtlPeerStatsRefuseCalls (1.3.6.1.2.1.10.21.1.2.2.1.6)

The protocol that a call uses can be found by associating the dial-peer entry (dialCtlPeerStatsEntry, 1.3.6.1.2.1.10.21.1.2.2.1 OID) in the DIAL-CONTROL-MIB with the corresponding dial-peer entry (cvVoIPPeerCfgEntry, 1.3.6.1.4.1.9.9.63.1.2.3.1 OID) in the CISCO-VOICE-DIAL-CONTROL-MIB. The cvVoIPPeerCfgSessionProtocol (1.3.6.1.4.1.9.9.63.1.2.3.1.1) OID in the CISCO-VOICE-DIAL-CONTROL-MIB uses a value of “Cisco (2)” for H.323 and “sip (3)” for SIP.

The CISCO-SIP-UA-MIB provides information on SIP call success/failure using the following OIDs:

Success

- cSipStatsSuccess (1.3.6.1.4.1.9.9.152.1.2.2)

- cSipStatsRedirect 1.3.6.1.4.1.9.9.152.1.2.3

Failure

- cSipStatsErrClient 1.3.6.1.4.1.9.9.152.1.2.4 (4xx errors)

- cSipStatsErrServer 1.3.6.1.4.1.9.9.152.1.2.5 (5xx errors)

- cSipStatsGlobalFail 1.3.6.1.4.1.9.9.152.1.2.6 (6xx errors)

Retry/Timeouts

- cSipStatsRetry 1.3.6.1.4.1.9.9.152.1.2.8 (retries/timeouts)

Table 14. CISCO-SIP-UA-MIB MIB Fields for 4xx, 5xx and 6xx SIP Responses

OID

OID#

SIP 4xx Error

cSipStatsClientBadRequestIns

1.3.6.1.4.1.9.9.152.1.2.4.1

400

cSipStatsClientBadRequestOuts

1.3.6.1.4.1.9.9.152.1.2.4.2

400

cSipStatsClientUnauthorizedIns

1.3.6.1.4.1.9.9.152.1.2.4.3

401

cSipStatsClientUnauthorizedOuts

1.3.6.1.4.1.9.9.152.1.2.4.4

401

cSipStatsClientPaymentReqdIns

1.3.6.1.4.1.9.9.152.1.2.4.5

402

cSipStatsClientPaymentReqdOuts

1.3.6.1.4.1.9.9.152.1.2.4.6

402

cSipStatsClientForbiddenIns

1.3.6.1.4.1.9.9.152.1.2.4.7

403

cSipStatsClientForbiddenOuts

1.3.6.1.4.1.9.9.152.1.2.4.8

403

cSipStatsClientNotFoundIns

1.3.6.1.4.1.9.9.152.1.2.4.9

404

cSipStatsClientNotFoundOuts

1.3.6.1.4.1.9.9.152.1.2.4.10

404

cSipStatsClientMethNotAllowedIns

1.3.6.1.4.1.9.9.152.1.2.4.11

405

cSipStatsClientMethNotAllowedOuts

1.3.6.1.4.1.9.9.152.1.2.4.12

405

cSipStatsClientNotAcceptableIns

1.3.6.1.4.1.9.9.152.1.2.4.13

406

cSipStatsClientNotAcceptableOuts

1.3.6.1.4.1.9.9.152.1.2.4.14

406

cSipStatsClientProxyAuthReqdIns

1.3.6.1.4.1.9.9.152.1.2.4.15

407

cSipStatsClientProxyAuthReqdOuts

1.3.6.1.4.1.9.9.152.1.2.4.16

407

cSipStatsClientReqTimeoutIns

1.3.6.1.4.1.9.9.152.1.2.4.17

408

cSipStatsClientReqTimeoutOuts

1.3.6.1.4.1.9.9.152.1.2.4.18

408

cSipStatsClientConflictIns

1.3.6.1.4.1.9.9.152.1.2.4.19

409

cSipStatsClientConflictOuts

1.3.6.1.4.1.9.9.152.1.2.4.20

409

cSipStatsClientGoneIns

1.3.6.1.4.1.9.9.152.1.2.4.21

410

cSipStatsClientGoneOuts

1.3.6.1.4.1.9.9.152.1.2.4.22

410

cSipStatsClientLengthRequiredIns

1.3.6.1.4.1.9.9.152.1.2.4.23

411

cSipStatsClientLengthRequiredOuts

1.3.6.1.4.1.9.9.152.1.2.4.24

411

cSipStatsClientReqEntTooLargeIns

1.3.6.1.4.1.9.9.152.1.2.4.25

413

cSipStatsClientReqEntTooLargeOuts

1.3.6.1.4.1.9.9.152.1.2.4.26

413

cSipStatsClientReqURITooLargeIns

1.3.6.1.4.1.9.9.152.1.2.4.27

414

cSipStatsClientReqURITooLargeOuts

1.3.6.1.4.1.9.9.152.1.2.4.28

414

cSipStatsClientNoSupMediaTypeIns

1.3.6.1.4.1.9.9.152.1.2.4.29

415

cSipStatsClientNoSupMediaTypeOuts

1.3.6.1.4.1.9.9.152.1.2.4.30

415

cSipStatsClientBadExtensionIns

1.3.6.1.4.1.9.9.152.1.2.4.31

420

cSipStatsClientBadExtensionOuts

1.3.6.1.4.1.9.9.152.1.2.4.32

420

cSipStatsClientTempNotAvailIns

1.3.6.1.4.1.9.9.152.1.2.4.33

480

cSipStatsClientTempNotAvailOuts

1.3.6.1.4.1.9.9.152.1.2.4.34

480

cSipStatsClientCallLegNoExistIns

1.3.6.1.4.1.9.9.152.1.2.4.35

481

cSipStatsClientCallLegNoExistOuts

1.3.6.1.4.1.9.9.152.1.2.4.36

481

cSipStatsClientLoopDetectedIns

1.3.6.1.4.1.9.9.152.1.2.4.37

482

cSipStatsClientLoopDetectedOuts

1.3.6.1.4.1.9.9.152.1.2.4.38

482

cSipStatsClientTooManyHopsIns

1.3.6.1.4.1.9.9.152.1.2.4.39

483

cSipStatsClientTooManyHopsOuts

1.3.6.1.4.1.9.9.152.1.2.4.40

483

cSipStatsClientAddrIncompleteIns

1.3.6.1.4.1.9.9.152.1.2.4.41

484

cSipStatsClientAddrIncompleteOuts

1.3.6.1.4.1.9.9.152.1.2.4.42

484

cSipStatsClientAmbiguousIns

1.3.6.1.4.1.9.9.152.1.2.4.43

485

cSipStatsClientAmbiguousOuts

1.3.6.1.4.1.9.9.152.1.2.4.44

485

cSipStatsClientBusyHereIns

1.3.6.1.4.1.9.9.152.1.2.4.45

486

cSipStatsClientBusyHereOuts

1.3.6.1.4.1.9.9.152.1.2.4.46

486

cSipStatsClientReqTermIns

1.3.6.1.4.1.9.9.152.1.2.4.47

487

cSipStatsClientReqTermOuts

1.3.6.1.4.1.9.9.152.1.2.4.48

487

cSipStatsClientNoAcceptHereIns

1.3.6.1.4.1.9.9.152.1.2.4.49

488

cSipStatsClientNoAcceptHereOuts

1.3.6.1.4.1.9.9.152.1.2.4.50

488

cSipStatsClientBadEventIns

1.3.6.1.4.1.9.9.152.1.2.4.51

489

cSipStatsClientBadEventOuts

1.3.6.1.4.1.9.9.152.1.2.4.52

489

cSipStatsClientSTTooSmallIns

1.3.6.1.4.1.9.9.152.1.2.4.53

422

cSipStatsClientSTTooSmallOuts

1.3.6.1.4.1.9.9.152.1.2.4.54

422

cSipStatsClientReqPendingIns

1.3.6.1.4.1.9.9.152.1.2.4.55

491

cSipStatsClientReqPendingOuts

1.3.6.1.4.1.9.9.152.1.2.4.56

491

cSipStatsServerIntErrorIns

1.3.6.1.4.1.9.9.152.1.2.5.1

500

cSipStatsServerIntErrorOuts

1.3.6.1.4.1.9.9.152.1.2.5.2

500

cSipStatsServerNotImplementedIns

1.3.6.1.4.1.9.9.152.1.2.5.3

501

cSipStatsServerNotImplementedOuts

1.3.6.1.4.1.9.9.152.1.2.5.4

501

cSipStatsServerBadGatewayIns

1.3.6.1.4.1.9.9.152.1.2.5.5

502

cSipStatsServerBadGatewayOuts

1.3.6.1.4.1.9.9.152.1.2.5.6

502

cSipStatsServerServiceUnavailIns

1.3.6.1.4.1.9.9.152.1.2.5.7

503

cSipStatsServerServiceUnavailOuts

1.3.6.1.4.1.9.9.152.1.2.5.8

503

cSipStatsServerGatewayTimeoutIns

1.3.6.1.4.1.9.9.152.1.2.5.9

504

cSipStatsServerGatewayTimeoutOuts

1.3.6.1.4.1.9.9.152.1.2.5.10

504

cSipStatsServerBadSipVersionIns

1.3.6.1.4.1.9.9.152.1.2.5.11

505

cSipStatsServerBadSipVersionOuts

1.3.6.1.4.1.9.9.152.1.2.5.12

505

cSipStatsServerPrecondFailureIns

1.3.6.1.4.1.9.9.152.1.2.5.13

580

cSipStatsServerPrecondFailureOuts

1.3.6.1.4.1.9.9.152.1.2.5.14

580

cSipStatsGlobalBusyEverywhereIns

1.3.6.1.4.1.9.9.152.1.2.6.1

600

cSipStatsGlobalBusyEverywhereOuts

1.3.6.1.4.1.9.9.152.1.2.6.2

600

cSipStatsGlobalDeclineIns

1.3.6.1.4.1.9.9.152.1.2.6.3

603

cSipStatsGlobalDeclineOuts

1.3.6.1.4.1.9.9.152.1.2.6.4

603

cSipStatsGlobalNotAnywhereIns

1.3.6.1.4.1.9.9.152.1.2.6.5

604

cSipStatsGlobalNotAnywhereOuts

1.3.6.1.4.1.9.9.152.1.2.6.6

604

cSipStatsGlobalNotAcceptableIns

1.3.6.1.4.1.9.9.152.1.2.6.7

606

cSipStatsGlobalNotAcceptableOuts

1.3.6.1.4.1.9.9.152.1.2.6.8

606

2.4.3.5 Transcoding Session Capacity and DSP Utilization

Real-time call statistics for transcoding sessions, and the DSPs used by transcoding, are available in the CISCO-DSP-MGMT-MIB OIDs covered in this section, including:

Total Statistics

- Transcoding sessions configured

- Transcoding sessions used

- Transcoding session available (unused)

Per-Profile Transcoding Statistics

- Transcoding sessions configured

- Transcoding sessions used

- Transcoding session available (unused)

Note: The OIDs in this section are currently supported only on the Cisco ISR and AS5000 Series platforms.

Table 15. Transcoding Session Capacity and DSP Utilization MIB Information

OID

OID#

Use/Operation

cdspTotAvailTranscodeSess

1.3.6.1.4.1.9.9.86.1.7.1

Total of all transcoding sessions configured in all profiles.

cdspTotUnusedTranscodeSess

1.3.6.1.4.1.9.9.86.1.7.2

Total of all unused transcoding sessions across all configured profiles.

cdspTranscodeProfileMaxConfSess

1.3.6.1.4.1.9.9.86.1.6.3.1.2

Number of transcoding sessions configured for the DSP profile given in cdspTranscodeProfileId.

cdspTranscodeProfileMaxAvailSess

1.3.6.1.4.1.9.9.86.1.6.3.1.3

Number of transcoding sessions available for the DSP profile given in cdspTranscodeProfileId.

The currently active, or used, total transcoding session count is given by:

cdspTotAvailTranscodeSess – cdspTotUnusedTranscodeSess


The currently active, or used, transcoding session count per DSP profile is given by:

cdspTranscodeProfileMaxConfSess – cdspTranscodeProfileMaxAvailSess

2.4.3.6 MTP Session Capacity and DSP Utilization

Real-time call statistics for hardware (HW) MTP sessions are available in the CISCO-DSP-MGMT-MIB OIDs covered in this section, including:

Total Statistics

- MTP sessions configured

- MTP sessions used

- MTP session available (unused)

Per-Profile MTP Statistics

- MTP sessions configured

- MTP sessions used

- MTP session available (unused)

MTP functionality is independent of Cisco UBE and this information is available for all Cisco UCM MTP deployments on Cisco IOS routers.

Table 16. Transcoding Session Capacity and DSP Utilization MIB Information

OID

OID#

Use/Operation

cdspTotAvailMtpSess

1.3.6.1.4.1.9.9.86.1.7.3

Total of all HW MTP sessions configured in all profiles.

cdspTotUnusedMtpSess

1.3.6.1.4.1.9.9.86.1.7.4

Total of all unused HW MTP sessions across all configured profiles.

cdspMtpProfileMaxConfSoftSess

1.3.6.1.4.1.9.9.86.1.6.4.1.2

Number of SW MTP sessions configured for the profile given in cdspMtpProfileId.

cdspMtpProfileMaxConfHardSess

1.3.6.1.4.1.9.9.86.1.6.4.1.3

Number of HW MTP sessions configured for the DSP profile given in cdspMtpProfileId.

cdspMtpProfileMaxAvailHardSess

1.3.6.1.4.1.9.9.86.1.6.4.1.4

Number of HW MTP sessions available for the DSP profile given in cdspMtpProfileId.

The total configured software MTP session count can be calculated by summarizing all the per profile entries (cdspMtpProfileMaxConfSoftSess for each profile). The current number of SW MTP sessions active or in use can be seen from the following CLI.

router#sh dspfarm all
DSPFARM Configuration Information:
Admin State: DOWN, Oper Status: DOWN - Cause code: ADMIN_STATE_DOWN
Transcoding Sessions: 0(Avail: 0), Conferencing Sessions: 0 (Avail: 0)
Trans sessions for mixed-mode conf: 0 (Avail: 0), RTP Timeout: 600
Connection check interval 600 Codec G729 VAD: ENABLED
Total number of active session(s) 0, and connection(s) 0
Total number of DSPFARM DSP channel(s) 0
Dspfarm Profile Configuration
Profile ID = 10, Service = MTP, Resource ID = 1
Profile Description :
Profile Service Mode : Non Secure
Profile Admin State : DOWN
Profile Operation State : DOWN
Application : SCCP Status : NOT ASSOCIATED
Resource Provider : NONE Status : NONE
Number of Resource Configured : 10
Number of Resource Available : 10
Hardware Configured Resources : 0
Hardware Available Resources : 0
Software Resources : 10
Codec Configuration
Codec : g711ulaw, Maximum Packetization Period : 30

The currently active, or used, total HW MTP session count is given by:

cdspTotAvailMtpSess – cdspTotUnusedMtpSess

The currently active, or used, HW MTP session count per DSP profile is given by:

cdspMtpProfileMaxConfHardSess – cdspMtpProfileMaxAvailHardSess

2.4.4 Licensing and Call Admission Control

Cisco UBE licensing is not yet enforced and therefore cannot be monitored with SNMP. However, the cvCallVolConnMaxCallConnectionLicenese (1.3.6.1.4.1.9.9.63.1.3.8.3) OID in the CISCO-VOICE-DIAL-CONTROL-MIB MIB is defined to reflect licensing information (when it becomes available).

This OID reflects a value of 0 by default, unless call admission control is configured, in which case the value reflects the “high” setting of the corresponding “call threshold global total-calls” CLI. E.g. if “call threshold global total-calls low 10 high 100” is configured, the OID value is set to 100.

It is recommended that you set the “call threshold global total-calls” CLI to the licenses purchased for the Cisco UBE router. Doing this ensures that when licensing becomes enforced in future, the monitoring of call volumes—and call rejections when exceeded—is already designed into your network and does not suddenly alter call traffic patterns.

2.4.5 Voice Quality MIBs

Voice quality can be monitored by the packet loss, delay and jitter statistics given in the MIB and OID covered in this section. These basic metrics can be calculated and summarized into metrics such as ICPIF, MOS and R-factor scores by your NMS system.

Packet statistics are available in the following MIBs:

CISCO-VOICE-DIAL-CONTROL-MIB

CISCO-RTTMON-ICMP-MIB

CISCO-RTTMON-MIB

CISCO-RTTMON-RTP-MIB

The CISCO-VOICE-DIAL-CONTROL-MIB provides packet statistics both for currently active calls (real-time statistics) as well for historical trending (calls that are already completed).

Table 17. Voice Quality: CISCO-VOICE-DIAL-CONTROL-MIB

OID

OID#

New/Changed

Use/Operation

cvVoIPCallActiveTable

11.3.6.1.4.1.9.9.63.1.3.2

Baseline

Real-time voice quality statistics on currently active calls.

cvVoIPCallHistoryTable

1.3.6.1.2.1.10.63.1.4.2

Baseline

Historical voice quality statistics for already completed calls.

The IP RTTMON MIBs provide various levels of generic packet and transmission statistics based on IP SLA probes configured on the router (using the IP SLAs RTP-Based VoIP Operation feature).

Table 18. Voice Quality: RTTMON MIBs

MIB

OID

OID#

Use/Operation

CISCO-RTTMON-ICMP-MIB

rttMonLatestIcmpJitterAvgJitter

1.3.6.1.4.1.9.9.42.1.5.4.1.44

ICMP Jitter

CISCO-RTTMON-MIB

rttMonJitterStatsAvgJitter

1.3.6.1.4.1.9.9.42.1.3.5.1.62

UDP Jitter

CISCO-RTTMON-RTP-MIB

rttMonRtpStatsIAJitterDSAvg

1.3.6.1.4.1.9.9.42.1.3.6.1.5

RTP Jitter

2.5 SNMP Traps

There are currently no SNMP traps implemented for Cisco UBE.

2.6 Syslog Messages

Syslog is a method to collect messages from devices to a server running a syslog daemon. Logging to a central syslog server helps in aggregation of logs and alerts. Cisco devices can send their log messages to a Unix-style SYSLOG service. A SYSLOG service simply accepts messages, and stores the messages in files or prints according to a simple configuration file. These messages are useful in routine troubleshooting and in incident handling.

Cisco devices have literally thousands of different messages that are sent to a central server (at the customer site) when an identified event occurs in the network. Events range from catastrophic (priority 0) to informational (priority 6).

The syslog daemon handles the recording of syslog messages and events in log files. The syslog message is composed of two main parts:

Header: Contains the date and time information along with the IP address or the computer name from which the message has originated.

Message: Includes the program or subsystem name and the message. The program or subsystem name and the message are separated by a colon.

The following is a summary of voice and call related Syslog message categories. Further information on individual messages within these categories can be found on Cisco.com in the Cisco IOS System Messages documentation.

CALL_CONTROL Messages

CALL_MGMT Messages

CALLRECORD Messages

CALLTREAT Messages

CALLTREAT_NOSIGNAL Messages

CCH323 Messages

CCM Messages

CSM Messages

CSM_TGRM Messages

CSM_TRUNK_MGR Messages

CSM_VOICE Messages

DSMP Messages

DSP_CONN Messages

DSPDUMP Messages

DSPFARM Messages

DSPRM Messages

FLEX_DNLD Messages

FLEXDSPRM Messages

GK Messages

HWCONF Messages

HWECAN Messages

IVR Messages

IVR_MSB Messages

IVR_NOSIGNALING Messages

PVDM Messages

PVDM2 Messages

PVDM2_DM Messages

PVDMPWR Messages

SIP Messages

VOICE_CODEC Messages

VOICE_ELOG Messages

VOICE_FILE_ACCT Messages

VOICE_IEC Messages

VOICE_RC Messages

VOICE_UTIL Messages

VOIPAAA Messages

VOIPFIB Messages

VOIP_RTP Messages

VTSP Messages

Table 19. Syslog Error Message Severity Levels

Level

Description

System Impact

0

Emergency

System unusable

1

Alert

Immediate action needed

2

Critical

Critical condition

3

Error

Error condition

4

Warning

Warning condition

5

Notification

Normal but significant condition

6

Informational

Informational message only

7

Debugging

Appears during debugging only

2.7 Embedded Event Manager (EEM)

The Cisco IOS Embedded Event Manager (EEM) is a unique subsystem within Cisco IOS Software. EEM is a powerful and flexible tool to automate tasks and customize the behavior of Cisco IOS Software and the operation of the device. You can use EEM to create and run programs or scripts directly on a router or switch. The scripts are referred to as EEM policies and can be programmed using a simple command-line-interface (CLI)-based interface or using a scripting language called Tool Command Language (Tcl). EEM allows you to harness the significant intelligence within Cisco IOS Software to respond to real-time events, automate tasks, create customer commands, and take local automated action based on conditions detected by the Cisco IOS Software itself.

More information is available at www.cisco.com/go/eem.

2.7.1 SIP Trunk Status

One example use of EEM for Cisco UBE is to monitor SIP trunk up/down status (based on SIP Out-of-Dialog Options ping) and generate a syslog message and an SNMP trap when a change is detected. Please note this is given merely as a guideline example and may require changes or adjustments based on your platform or software release.

The relevant configuration is:

! Note: The number (10) in the “track” statement below must match
! the dial-peer number
track 10 stub-object
!
dial-peer voice 10 voip
destination-pattern .T
voice-class sip options-keepalive
session protocol sipv2
session target ipv4:10.x.x.x
session transport tcp
codec g711ulaw
!
event manager environment dial_peer_number 10
event manager environment check_interval 30
event manager directory user policy "flash:/"
event manager applet siptrunk_down
event track 10 state down
action 10 snmp-trap strdata "siptrunk DOWN"
action 20 syslog msg "siptrunk down"
event manager policy check_dial_peer_status.tcl

Example text for the Tcl script (flash:check_dial_peer_status.tcl) is:

::cisco::eem::event_register_timer watchdog time $check_interval nice 1
#
# Namespace imports
#
namespace import ::cisco::eem::*
namespace import ::cisco::lib::*
#--- Check required environment variable(s) has been defined
if {![info exists dial_peer_number]} {
set result "EEM Policy Error: variable dial_peer_number has not been set"
error $result $errorInfo
}
#------------------- " cli open" -------------------
if [catch {cli_open} result] {
error $result $errorInfo
} else {
array set cli $result
}
#----------------------- "enable" ----------------------
if [catch {cli_exec $cli(fd) "enable"} result] { error $result $errorInfo }
#-------------- grab sip ood options-ping and track status --------------
if [catch {cli_exec $cli(fd) "show dial-peer voice $dial_peer_number | inc options-keepalive dial-peer action"} result] {
error $result $errorInfo
}
set cmd_output $result
if [catch {cli_exec $cli(fd) "show track $dial_peer_number | inc State"} result] { error $result $errorinfo
}
set track_state $result
#-------------- set stub status --------------
if [string match "*busyout*" $cmd_output] {
if [string match "*Up*" $track_state] {
if [catch {cli_exec $cli(fd) "conf t" } result] { error $result $errorInfo }
if [catch {cli_exec $cli(fd) "track $dial_peer_number stub-object" } result]
{ error $result $errorInfo }
if [catch {cli_exec $cli(fd) "default-state down" } result]
{ error $result $errorInfo }
if [catch {cli_exec $cli(fd) "end" } result] { error $result $errorInfo }
}
}
if [string match "*active*" $cmd_output] {
if [string match "*Down*" $track_state] {
if [catch {cli_exec $cli(fd) "conf t" } result] { error $result $errorInfo }
if [catch {cli_exec $cli(fd) "track $dial_peer_number stub-object" } result]
{ error $result $errorInfo }
if [catch {cli_exec $cli(fd) "default-state up" } result]
{ error $result $errorInfo }
if [catch {cli_exec $cli(fd) "end" } result] { error $result $errorInfo }
}
}
#--------------------- cli close ------------------------
if [catch {cli_close $cli(fd) $cli(tty_id)} result] {
error $result $errorInfo
}

2.8 IP SLA

Cisco IOS IP Service Level Agreements (SLAs) is a network performance measurement and diagnostics tool that uses active monitoring, which generates traffic in a reliable and predictable manner to measure network performance.

A summary of IP SLA capabilities is given below. More information is available at www.cisco.com/go/ipsla.

Table 20. Cisco IOS IP SLAs Operations and Applications

IP SLA

Measurement Capability

Key Applications

RTP-Based VoIP

Interarrival jitter
Estimated R factor
MOS-CQ
Round-trip time (RTT) latency
Packet loss
Packets missing in action
One-way latency
Frame loss
MOS-LQ (destination-to-source)

Networks that carry voice and video traffic

UDP Jitter for VoIP

Round-trip delay, one-way delay, one-way jitter, one-way packetloss
VoIP codec simulation G.711 μ-law, G.711 a-law, and G.729A
MOS and ICPIF voice quality scoring capability
One-way delay requires time synchronization between theCisco IOS IP SLAs source and target routers

Most common operations for networks that carry voice traffic, such as IP backbones

UDP Echo

Round-trip delay

Accurate measurement of response time of UDP traffic

UDP Jitter

Round-trip delay, one-way delay, one-way jitter, one-way packetloss
One-way delay requires time synchronization between theCisco IOS IP SLAs source and target routers

Most common operations for networks that carry voice or video traffic, such as IP backbones

TCP Connect

Connection Time

Server and application performance monitoring

Domain Name System (DNS)

DNS Lookup Time

DNS performance monitoring, troubleshooting

Dynamic Host Configuration Protocol (DHCP)

Round-trip time to get an IP address

Response time to a DHCP server

Internet Control Message Protocol (ICMP) Echo

Round-trip delay

Troubleshooting and availability measurement

ICMP Path Echo

Round-trip delay for the full path

Troubleshooting

ICMP Path Jitter

Round-trip delay, jitter and packet loss for the full path

Troubleshooting

2.9 NetFlow

Cisco IOS NetFlow efficiently provides a key set of services for IP applications, including network traffic accounting, usage-based network billing, network planning, security, Denial of Service monitoring capabilities, and network monitoring. NetFlow provides valuable information about network users and applications, peak usage times, and traffic routing.

More information is available at www.cisco.com/go/netflow.


3 Supported Management Applications

The following table provides information on Management Applications that can be used to manage Cisco UBE.

Table 21. Supported Management Applications

Management Application

Applicable Operations Phase(s)

Application Description

Support

Website

Cisco Applications

Cisco Configuration Professional (CCP)

Staging, Provisioning, On-going changes

A GUI device management tool for Cisco IOS®Software-based access routers. It simplifies router, firewall, IPS, VPN, unified communications, WAN, and basic LAN configuration through easy-to-use wizards

Yes (v2.3)

http://www.cisco.com/go/ciscocp

Cisco Configuration Engine (CCE)

Staging, Deployment

A network management software solution that provides a highly distributive delivery system for configuration updates and device image upgrades to thousands of devices

Yes (as a router)

http://www.cisco.com/go/ciscoce

Cisco Works LMS

Monitoring, Troubleshooting, Change Management

Simplifies the configuration, administration, monitoring, and troubleshooting of Cisco networks

Yes (as a router)

http://www.cisco.com/go/lms

Cisco Works NCM

Change Management

Supporting configuration of a new product/feature

No

Cisco Security Manager (CSM)

Staging, Monitoring, Troubleshooting for Security

Security management (configuration and event management) across a wide range of Cisco security appliances

No

Cisco Unified Operations Manager (CUOM)

Monitoring and Troubleshooting for Voice

Features out-of-the-box, real-time, service-level monitoring of all system elements. It performs automatic discovery for the entire system and provides diagnostics for rapid troubleshooting

Yes (v2.2)

http://www.cisco.com/go/cuom

Cisco Unified Provisioning Manager (CUPM)

Staging for Voice

Supports the implementation of Cisco Unified Communications, as well as ongoing, simplified operational provisioning and activation services for individual subscriber changes

Future

http://www.cisco.com/go/cupm

Cisco Unified Service Monitor (CUSM)

Monitoring for Voice

Monitors active calls supported by the Cisco Unified Communications System and provides near real-time notification when the voice quality of a call, fails to meet a user-defined quality threshold

No

http://www.cisco.com/go/cusm

CiscoWorks QoS Policy Manager

Staging and Monitoring for QoS

Supports centralized management of network quality of service (QoS) as well as QoS provisioning and monitoring capabilities

Yes

http://www.cisco.com/en/US/products/sw/cscowork/ps2064/index.html

Cisco License Manager (CLM)

Staging for IOS License, Change Management

ManagesCisco IOS Software activationand license management for a wide range of Cisco platforms running IOS as well as other operating systems

Yes

http://www.cisco.com/go/clm

3rd Party Applications

CA eHealth

Performance Monitoring, Reporting

A performance management solution that ensures quality of service across your entire infrastructure

No

http://www.ca.com

CA Spectrum

Fault Monitoring

An integrated management solution for business service management, fault isolation and root cause analysis, and network configuration management

No

http://www.ca.com

EMC Smarts

Performance Monitoring, Reporting

Automates real-time analysis of network connectivity problems and provides the critical lead time needed to address network performance problems

No

http://www.emc.com

EMC Voyence

Change Management

Automates the entire configuration management lifecycle, including Design, Change and Compliance

No

http://www.emc.com

Arcana iManage

Provisioning, Monitoring, Change Management

A provisioning and monitoring solution to help companies deploy advanced services based on the Cisco Integrated Services Router and Unified Communications platforms

Yes

http://www.arcananet.com

Solarwinds

Fault and Performance Monitoring

Offers network managers a comprehensive and easy-to-understand view of network health—from fault and performance monitoring to configuration and IP address management

Yes

http://www.solarwinds.com

InfoVista

Performance Monitoring

A network performance managementand service assurance solution that enables telcos and enterprises to effectively meet and exceed performance expectations and service-level guarantees for today’s communications technologies and services

No

http://www.infovista.com


4 Management Recommendations

All general Cisco IOS router management (provisioning, monitoring and troubleshooting) methods and information are applicable to Cisco UBE.

4.1 Provisioning Recommendations

Cisco UBE can be provisioned using either the Cisco IOS router CLI or by using a graphical provisioning management application.

4.1.1 Command Line (CLI)

General provisioning of Cisco UBE is done via Cisco IOS CLI. The Cisco UBE IOS CLI Configuration Guide can be found at www.cisco.com/go/cube > Configure > Configuration Guides > Cisco Unified Border Element with Gatekeeper Configuration Guide.

Note that on an ISR Generation 2 (Cisco 2900 and 3900 series platforms), the following CLI is required to enable the Cisco UBE features:

voice service voip
mode border-element

4.1.2 Graphical (GIU)

Graphical (GUI) provisioning of Cisco UBE can be done by using Cisco Configuration Professional (CCP) 2.3 or later. General information on CCP can be found at www.cisco.com/go/ccp.

A service provider SIP trunk configuration template is provided in CCP 2.3, as shown below. The drop-down values under “Generic” provides specific service provider selection and accelerates the Cisco UBE configuration necessary to connect to the select service provider. Service providers not yet explicitly supported in the drop-down can be configured using the “Generic” template.

A summary of Cisco UBE features supported by CCP 2.3 include the features represented by the following CLI segments.

Global:

call threshold global total-calls low 7920 high 9000
ip domain name mydomain.com
ip name-server 10.25.135.23

Global VoIP Services:

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
address-hiding
supplementary-service h450.12
sip
outbound-proxy ipv4:5.5.5.5
early-offer forced
midcall-signaling passthru
header-passing error-passthru
g729 annexb-all
asserted-id pai

SIP UA:

sip-ua
sip-server ipv4:2.2.2.2
remote-party-id
registrar dns:cisoc.com
authentication username ciscocp password ciscocp
credentials username test password test realm test

Translation Rules:

voice translation-rule 2
rule 1 /12345/ /4083/
voice translation-profile test
translate called 2
translate calling 2
translate redirect-called 2

VoIP Dial Peer:

dial-peer voice 1 voip
description xxx
corlist incoming test
preference 5
session target sip-server
answer-address 408
destination-pattern 1234
session protocol sipv2
incoming called-number .T
voice-class codec 5
corlist outgoing test
dtmf-relay cisco-rtp
translation-profile outgoing test
translation-profile incoming test
codec g711alaw

Voice Class Codec:

voice class codec 5
codec preference 1 g711ulaw
codec preference 2 g729r8

Class of Restriction (COR):

dial-peer cor custom
name international
dial-peer cor list test
member international

4.2 SIP Trunk Security Recommendations

Like any other network and router device, Cisco UBE should be locked down against security attacks. Cisco UBE offers a variety of features to mitigate a range of attacks in the following categories:

Denial of Service (DOS): Overrunning the system with traffic

Identify and Service Theft: A rogue endpoint masquerading a legitimate endpoint and thereby using network services such as making toll calls

Privacy: Unauthorized listening to, or recording of, calls

4.2.1 Service Provider (SP) SIP Trunk Security

When Cisco UBE is deployed in SP SIP trunk configurations, one of its major functions is to serve as a security point handing off enterprise traffic to the SP network. To ensure security for a SIP trunk deployment, the following features must be configured at a minimum:

Access Lists (ACLs) to Allow/Deny Explicit Sources of Calls: Permit traffic only from the service provider SBC on the outside, and only the valid call agent(s) on the inside of the network. No other endpoint or source should be able to make or receive calls to Cisco UBE.

CAC to Limit Call Arrival Rates and Max Active Calls: Deploy total call limits, per dial-peer call limits, call spike detection and CPU protection against potential SIP DOS attacks.

Toll Fraud Lock-Down: Ensure that only legitimate endpoints can make authorized toll calls via Cisco UBE.

Additional optional features may be configured as needed:

SIP Listen Port: Change the default 5060 SIP port to another port number.

voice service voip
sip
shutdown
voice service voip
sip
listen-port non-secure 2000 secure 2050

SIP Registration: A SP SIP trunk requiring a registration sequence is more secure than one that doesn’t. However, many SPs do not currently support or offer SIP registration.

sip-ua
credentials username 1001 password 0822455D0A16 realm cisco.com

SIP Digest Authentication: Cisco UBE responds to SIP Digest Authentication challenges from a SP call agent.

sip-ua
authentication username xxx password yyy

SIP Hostname Validation: In addition to ACLs, this configuration can further limit the sources of traffic accepted by Cisco UBE.

sip-ua
permit hostname dns:example1.sip.com
permit hostname dns:example2.sip.com
permit hostname dns:10.10.10.10

PPI/PAI: SP SIP trunks that offer P-Preferred-Identity (PPI) or P-Asserted-Identity (PAI) as per RFC3325 are more secure than those that do not. Configure these features if they are available from your SIP trunk SP. These features can be configured either at the global level or per dial-peer as given below. If the SIP trunk SP and your enterprise call agent both support these headers, then Cisco UBE can just pass them through. If your call agent does not support them, or you wish to translate one header type to another before handing them off to the SP, then use these features on Cisco UBE to do so.

voice service voip
sip
asserted-id pai
dial-peer voice 100 voip
voice-class sip asserted-id pai
voice service voip
sip
asserted-id ppi
dial-peer voice 100 voip
voice-class sip asserted-id ppi

Firewall: The IOS Firewall may be collocated with Cisco UBE on the same router to provide IP protection for non-SIP traffic if an external firewall is not already deployed.

Tcl: Tcl scripting applications can be configured on Cisco UBE dial-peers to do additional security checks before allowing or denying a call. Checks can be done against a short list of numbers held locally in router memory, or checks can be done against an external server or database. In this way, whitelist/blacklist applications can be built. Tcl scripts can also be written to query the caller for a PIN or authorization code before allowing the call.

Encryption: Signaling (TLS) and media (Secure RTP, SRTP) encryption: SP SIP trunks providers do not offer TLS/SRTP encryption today, but may do so in future. Cisco UBE supports TLS to non-TLS connections and SRTP.

Monitor CDR: Whether or not actual accounting information is required, it is recommended to monitor CDR from Cisco UBE to scan for call patterns and volumes that may indicate unauthorized use. Some toll fraud hackers bypass the enterprise call agent for the very purpose of not having their calls show up in the call agent CDR/billing records and instead address their fraudulent calls directly to the PRI gateway or SBC—therefore monitoring CDR from the gateway/SBC itself is important to see these call patterns. Also for call patterns where the hacker is on the PSTN and hairpins a call through your PRI gateway or SBC back out to the PSTN—these calls never hit your call agent and do not show up in call agent CDR.

4.2.2 Toll Fraud Security

Toll fraud is still the most prevalent security issue for devices that provide access to the PSTN, be they traditional PRI gateways or an SBC like Cisco UBE anchoring a PSTN SIP trunk. To protect against PSTN toll fraud, ensure the following features already discussed in the previous section are configured:

ACLs

SIP listen port change

Tcl

CDR

In addition to the above, it is recommended to configure:

Explicit Incoming and Outgoing Dial-Peers: The more explicit you can make the “incoming called-number” (for an incoming dial-peer) or “destination-pattern” (outgoing dial-peer) the more secure it is. Avoid use of default incoming dial-peer 0 which is promiscuous and allows all incoming connections.

Trunk Access Codes Using Translation Rules: Protect calls to expensive PSTN destinations or undesirable locations (perhaps international calls, calls to certain countries, etc.) with trunk access codes in front of the PSTN direct dial string. These codes can be transparent to your legitimate user base by inserting the code at your call agent (e.g. 89923 for calls to country-X) and deleting the code at Cisco UBE before passing the call to the PSTN. The use of this precludes a hacker directly addressing the SIP trunk and dialing direct to expensive locations (while bypassing your call agent).

Close Unused H.323/SIP Ports and Transport Mechanisms: By default these ports are open when a voice-enabled software load is deployed on the router (either as a PRI gateway or Cisco UBE).

sip-ua
no transport tcp
no transport udp

4.2.3 New Security Operation in Cisco IOS 15.1.2T

To help mitigate toll fraud opportunities, as of 15.1.2T Cisco IOS no longer allows connections from “unknown” sources to connect by default. Only sources on the IP Trust List are allowed (by default) and all other calls are rejected.

IP addresses defined in the “session target ipv4:” commands on dial-peers are automatically included in the IP Trust List. Additional valid source IP addresses can be added manually to the Trust List if needed by using the following CLI:

voice service voip
ip address trusted list
ipv4 20.20.20.1

While it is recommended to use the increased security operation available in 15.1.2T, pre-15.1.2T IOS operation can be restored by using the CLI:

no ip address trusted authenticate

4.3 Monitoring Recommendations

The following aspects of Cisco UBE are recommended to be monitored.

Router Inventory and Health

Interface Health

SIP Trunk Status

Call Traffic

- Trunk utilization and H.323/SIP Session Capacity

- Call arrival rate

- Call success/failure statistics

- SIP retries statistics

- Transcoding Session Capacity and DSP Utilization

- Media Termination Point (MTP) Session Capacity

Licensing and Call Admission Control

Voice Quality

Billing/CDR

The following table lists key “show” commands giving output that enables you to monitor Cisco UBE health, traffic and activity.

Table 22. Key “show” Commands on Cisco UBE

Category

Command

Information Provided

Configuration

show version

Displays the version of the image on the router

show flash:

Displays information about flash: file system

show ip interface brief

Displays brief summary of IP status and configuration

show startup-config

Displays the startup configuration on the router

show running-config

Displays the present/running con configuration on the router

show debug

Displays the debugs currently enabled

show voice iec desc <>

Displays definition of an Internal Error Code

show logging

Displays the contents of logging buffers

Traffic

show call active voice

Displays complete details of an active call like media settings, call statistics, SRTP on/off, etc.

show call active voice brief

Displays a brief version of active voice calls, e.g. transmitted and received packets and duration of call

show call active voice compact

Displays a compact version of active voice calls

show voip rtp connections

Displays active RTP connections

show call history voice

Displays calls stored in the history table for voice

Router Health

show processes cpu sorted <1min/5min/5sec>”

Displays sorted output based on percentage of CPU utilization

show processes cpu sorted history

Displays CPU history information in a graph format

show memory processor

Displays memory statistics

show process memory <>

Displays memory per process

show memory debug leaks

Runs the memory leak detector

show alignment

Displays alignment data and spurious memory references

CAC

show call threshold config

Displays configured resource information

show call treatment config

Displays call admission control information

show call treatment stats

Displays call treatment statistics

SIP

show sip-ua connections udp brief

Displays summary of SIP UDP connection information

show sip-ua connections udp detail

Displays details of SIP UDP connection information

show sip-ua connections tcp brief

Displays summary of SIP TCP connection information

show sip-ua connections tcp detail

Displays details of SIP TCP connection information

show sip-ua register status

Displays SIP registration status

Transcoding and DSPs

show diag

Displays diagnostic and hardware information for port adapters and modules

show sdspfarm units

Displays transcoder registration status

show sccp connection

Displays the active SCCP connections

show sccp

Displays SCCP protocol information

show dspfarm dsp active

Displays the active DSPs

show call active voice | inc CoderTypeRate=”

Displays call connectivity, codec and the media type information

show call active voice comp

Displays codec information for transcoding calls

DTMF Relay

show call active voice | inc tx_DtmfRelay

Displays the DTMF-relay used for the call

Security

show sip-ua connections tcp tls brief

Displays summary information on whether the transport used for the call is TLS or not

Show sip-ua connections tcp tls detail

Displays detailed information on whether the transport used for the call is TLS or not

Gatekeeper

show gatekeeper status

Displays if the Gatekeeper state is Up/Down

show gateway

Displays status of H.323 gateway

show gatekeeper endpoints

Displays E.164 endpoint register status

show gatekeeper calls

Displays current gatekeeper call status

show gatekeeper zone prefix all

Displays all zone and gateway registered E.164 prefixes

show gatekeeper gw-type-prefix

Displays gateway Technology Prefix Table

show gatekeeper zone status

Displays available bandwidth in ACF

show h323 gateway cause-codes

Displays H.323 disconnect cause codes

4.4 Troubleshooting Recommendations

4.4.1 General

The following procedure is recommended to collect usable debug information:

Configure “logging buffer 10000000” and “no logging console”

Enter the “clear logging” command

Perform the test


Collect the logs using following two commands

- term length 0

- show logging

The following table lists key “debug” commands giving output that enables you to troubleshoot problems on Cisco UBE.

Table 23. Key “debug” Commands on Cisco UBE

Category

Command

SIP and H.323

debug voip ccapi all

debug voip ccapi input

debug voip ccapi inout

debug voip dialpeer all

debug voip ipipgw

SIP

debug ccsip all

H.323

debug cch323 all

debug h225 asn

debug h225 events

debug h245 asn

debug h245 events

Transcoding and MTP

debug sccp all

debug sccp events

debug sccp messages

debug sccp errors

debug voip xcodemsp

Media

debug rtpspi error

debug voip rtp error

debug voip app

The following key “show” commands provide output that enables you to troubleshoot problems on Cisco UBE.

sh call history ?
fax Show calls stored in the history table for fax
media Show calls stored in the history table for media
video Show calls stored in the history table for video
voice Show calls stored in the history table for voice
!
sh voice call ?
<0-0> Voice interface slot #
status Show status for active calls
summary Summary of all voice calls
!
sh voip ?
debug Show voip debug info
rtp Display Real Time Protocol (RTP) information
!
sh voip rtp ?
connections Display all the active RTP connections
!
sh voice statistics ?
csr Show Call Statistics Records information
iec Show Internal Error Code information
interval-tag Show Voice Statistics time-range intervals
memory-usage Show current memory utilization of voice statistics

4.4.2 High-Traffic-Volume Troubleshooting (PCD)

Under high traffic conditions on an ISR G2 (e.g. a Cisco 3945 or 3945E) or an ASR (e.g. a Cisco ASR1006), the generic Cisco IOS debugging commands may be too verbose to allow effective debugging of a production Cisco UBE. Under these conditions the Per Call Debugging (PCD) tools can be used to minimize debug output to a particular call of interest.

PCD enables the configuration of debugging to be done to circular memory buffers rather than console output. Trigger conditions are set up to monitor the buffer contents and print out to the console only debug (from the memory buffers) that matches a particular trigger condition. Debug from the memory buffers can also be directed to an offline system for further analysis or interpretation.

Sample trigger points include:

SIP 4xx, 5xx and 6xx error messages

Q.850 cause codes

CAC limits

- Call treatment – cause code busy

- Call treatment – cause code noQos

- Call treatment – cause code no-resource

A summary of configuring and using PCD include the following steps:

Step 1: Define buffers and buffer sizes

per-call num-buffer <num>
per-call buffer-size debug <num>

Step 2: Turn per-call debugging on/off

per-call shutdown
per-call active debug
per-call inactive

Step 3: Set trigger points

per-call trigger cause 1
per-call trigger cause 41
per-call trigger sip-message 404
per-call trigger sip-message 488

Step 4: Export debug buffer content

per-call export primary [flash | ftp | http | pram | rcp | tftp] secondary [flash | ftp | http | pram | rcp | tftp]

Step 5: Show buffer content status

show per-call stat
show per-call buffer list
show per-call buffer content <buf-id>
show voice per-call trigger

Step 6: Show buffer contents on console

router#show per-call buffer content ?
<0-10000000> Specify the buffer number
router#show per-call buffer content 1

Buffers are indexed/referenced based on the GUID of the call:

router#show per-call buffer list
No. GUID Usage Last Updated Status
=====================================================================
0 490C81D9-34E0-11DE 73402 Apr 30 17:08:00.784 PCD_BUFFER_INUSE
8656-E29654047181
1 00000000-0000-0000 0 PCD_BUFFER_FREE
0000-000000000000
2 00000000-0000-0000 0 PCD_BUFFER_FREE
0000-000000000000
3 00000000-0000-0000 0 PCD_BUFFER_FREE
0000-000000000000
4 00000000-0000-0000 0 PCD_BUFFER_FREE
0000-000000000000

4.4.3 SIP Ladder Diagrams

The Cisco IOS IP Traffic Capture feature can be used to build protocol (SIP) ladder diagrams for protocol troubleshooting. This feature captures packets on an interface and builds a pcap file that can be copied to an offline system for protocol analysis by a tool such Wireshark (freeware, from www.wireshark.org).

A summary of capturing and analyzing SIP protocols information using these tools include the following steps:

Step 1: Configure a capture profile

! create profile
ip traffic-export profile TAC mode capture
bidirectional
incoming access-list 123
outgoing access-list 123
!
! access-list to filter only SIP messages (port 5060)
access-list 123 permit udp any any eq 5060
access-list 123 permit tcp any any eq 5060
!
! apply to an interface, default memory is 5M

interface fa0/0
ip traffic-export apply TAC [size <bytes>]

Step 2: Capture traffic with these exec (enable) level commands

router#traffic-export interface fa0/0 clear
router#traffic-export interface fa0/0 start
<capture the problem>
router#traffic-export interface fa0/0 stop

Note: The exec cmds don’t appear until a profile has been configured.

Step 3: Export the pcap file to a server

router#traffic-export interface fa0/0 copy ftp://x.x.x.x/capture.pcap

Step 4: Display a ladder diagram using Wireshark

Note: Allows filtering of calling/called numbers when creating the flow graph.

The IP Traffic Capture tools are currently available on the ISR G1 and G2 series platforms only.


5 Glossary

AAA: Authentication, Authorization, and Accounting

ACL: Access List

ASR: Aggregation Services Routers

CAC: Call Admission Control

CDR: Call Detail Record

CCE: Cisco Configuration Engine

CCP: Cisco Configuration Professional

CLI: Command Line Interface

CLM: Cisco License Manager

CUBE: Cisco Unified Border Element

CUCM: Cisco Unified Communications Manager

CUOM: Cisco Unified Operations Manager

DOS: Denial of Service

DSP: Digital Signal Processing

DTMF: Dual-tone Multi Frequency

EEM: Embedded Event Manager

FQDN: Fully Qualified Domain Name

GK: Gatekeeper

GUI: Graphical User Interface

GW: Gateway

HA: High Availability

HSRP: Hot Standby Router Protocol

ICPIF: Calculated Planning Impairment Factor

ISR: Integrated Services Router

ISR G2: Integrated Services Router Generation 2

LMS: LAN Management Solution

MIB: Management Information Base

MOS: Mean Opinion Score

MTP: Media Termination Point

NTP: Network Time Protocol

OID: Object identifier

OOD: Out of Dialog

PBX: Private Branch Exchange

PCD: Per Call Debugging

PDD: Post Dial Delay

PSTN: Public Switched Telephone Networks

QoS: Quality of Service

RADIUS: Remote Authentication Dial-in User Service

RAI: Resource Availability Indicator

RPID: Remote Party ID

RSVP: Resource Reservation Protocol

RTP: Real-time Protocol

SBC: Session Border Controller

SCCP: Skinny Client Control Protocol

SIP: Session Initiation Protocol

SLA: Service Level Agreement

SNMP: Simple Network Management Protocol

SP: Service Provider

SRTP: Secure RTP

Tcl: Tool Command Language

TCP: Transmission Control Protocol

TDM: Time Division Multiplexing

TLS: Transport Layer Security

UC: Unified Communications

UDP: User Datagram Protocol

UNI: User to Network Interface

URI: Universal Resource Identifier

VAD: Voice Activity Detection


6 References

Cisco UBE on Cisco.com

Cisco Communications Transformations Whitepapers > Section on Whitepapers

Cisco Interoperability Portal > Cisco Unified Border Element (CUBE)/SIP Trunking Solutions

Cisco UBE IOS Configuration Documentation

Cisco UBE IOS Configuration Application Notes and Examples Documentation

Cisco IOS Voice Command Reference

Cisco SRND Portal (CUCM and CVP SIP Trunk Documentation)

- CUCM 8.x SRND

- CUCM 7.x SRND

- CUCM 6.x SRND

- CVP 7.0 SIP Trunk Integration

Cisco.com MIB Locator tool

Cisco.com SNMP Object Navigator tool