Cisco Unified Border Element (CUBE) Management and Manageability Specification

Table of Contents

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

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

2       Embedded Management............................................................................................................ 5

2.1     CLI—Provisioning................................................................................................................... 5

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

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

2.1.3          H.323 CLI...................................................................................................................... 7

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

2.1.5          Security Features CLI.................................................................................................... 8

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       How to buy............................................................................................................................... 44

7       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 3^rd 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

 

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: https://www.cisco.com/c/en/us/products/unified-communications/unified-border-element/index.html> Configure > Configuration Guides > Cisco Unified Border Element Configuration Guide.

●   Cisco UBE IOS configuration is fully documented at: https://www.cisco.com/c/en/us/products/unified-communications/unified-border-element/index.html > 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 https://www.cisco.com/c/en/us/products/unified-communications/unified-border-element/index.html > Configure > Configuration Guides > Configuration Examples and TechNotes

●   Cisco UBE interoperability configuration guides with service provider SIP trunks and 3^rd 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 https://www.cisco.com/c/en/us/products/unified-communications/unified-border-element/index.html > 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

 

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

Table 3.       CISCO-MEMORY-POOL-MIB

 

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

Table 4.       IF-MIB

 

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

 

Table 6.       CISCO IOS MIBs that Contain Historical Call Information

 

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

 

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

 

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

 

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

 

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

 

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

 

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

 

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

 

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

 

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

 

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

 

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

 

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

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 https://www.cisco.com/c/en/us/products/ios-nx-os-software/ios-embedded-event-manager-eem/index.html.

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

 

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

 

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 https://www.cisco.com/c/en/us/products/unified-communications/unified-border-element/index.html > 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 https://community.cisco.com/t5/cisco-insider-user-group/ct-p/ccp-home.

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, allowed list/blocked list 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

 

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

 

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          How to buy

To view buying options and speak with a Cisco sales representative, visit https://www.cisco.com/c/en/us/buy.html.

7          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