Unified Border Element (CUBE) with Cisco Unified Communications Manager (CUCM) Configuration Example

Document ID: 99863
Introduction Prerequisites Requirements Components Used Conventions Configure Configurations SIP User Agent Configuration Interconnecting with Cisco Unified Communications Manager Transcoding on the Cisco Unified Border Element Using Tcl IVR on the Cisco Unified Border Element Full Sample Configuration Verify Troubleshoot Troubleshooting Commands Related Information

Introduction
The Cisco Unified Border Element facilitates simple and cost−effective connectivity between enterprise unified communications Session Initiation Protocol (SIP) trunks to the public−switched telephone network (PSTN). Designed to meet enterprise and service−provider Session Border Controller (SBC) device needs, the Cisco Unified Border Element (CUBE) is an integrated Cisco IOS® Software application that runs on: • Cisco 2800 Series Integrated Services Routers • Cisco 3800 Series Integrated Services Routers • Cisco 2600XM Series Multiservice Platforms • Cisco 3700 Series Routers • Cisco 7200VXR Routers • Cisco 7301 Routers • Cisco AS5400XM and AS5350XM Access Gateways Direct IP interconnections between unified communications networks offer greater flexibility to support emerging services when compared with traditional public−switched telephone network (PSTN) time−division multiplexing (TDM) interconnections. The Cisco Unified Border Element provides a network−to−network interface point for: • Signaling interworkingH.323, SIP • Media interworkingdual−tone multifrequency [DTMF], fax, modem, and codec transcoding • Address and port translationsprivacy and topology hiding • Billing and call detail record (CDR) normalization • Quality−of−service (QoS) and bandwidth managementQoS marking using differentiated services code point [DSCP] or type of service (ToS), bandwidth enforcement using Resource Reservation

Prerequisites Requirements There are no specific requirements for this document. All of the devices used in this document started with a cleared (default) configuration. make sure that you understand the potential impact of any command. and call−control servers in many different application environments. you are presented with the information to configure the features described in this document. and codec .323 to SIP or SIP to SIP. If your network is live. Configurations This configuration enables the basic Cisco Unified Border Element functionality on a platform. This functionality terminates an incoming VoIP call and re−originates it with the use of an outbound VoIP dial−peer. from advanced enterprise voice and/or video services with Cisco Unified Communications Manager or Cisco Unified Communications Manager Express.323 enterprise unified communications networks. Configure In this section. DTMF type. voice service voip allow−connections h323 to sip allow−connections sip to h323 allow−connections sip to sip allow−connections h323 to h323 Configure the incoming and outgoing dial−peers with the relevant protocol. The calls can be H. The Cisco Unified Border Element is used by enterprise and small and medium−sized organizations to interconnect SIP PSTN access with SIP and H. Conventions Refer to the Cisco Technical Tips Conventions for more information on document conventions. The Cisco Unified Border Element provides organizations with all the border controller functions integrated into the network layer to interconnect unified communications voice and video enterprise−to−service−provider architectures. The information in this document was created from the devices in a specific lab environment. IP phones. as well as simpler toll bypass and voice over IP (VoIP) transport applications. Note: Use the Command Lookup Tool ( registered customers only) to obtain more information on the commands used in this section. Components Used The information in this document is based on the Cisco Unified Border Element (CUBE).Protocol [RSVP] and codec filtering A Cisco Unified Border Element interoperates with many different network elements including voice gateways.

H.13..1 or later. which include the Cisco 2800 and 3800 series ISRs).323 Trunk to the Cisco Unified Border Element There are two methods of defining an H. which also requires an MTP.com authentication username xyz password xyz realm cisco. Cisco Unified Communications Manager must be configured for inbound and outbound H.323 to H.323 trunk to the Cisco Unified Border Element on the Cisco Unified Communications Manager: • With a gatekeeperConfigure an H.. SIP User Agent (UA) sip−ua registrar ipv4:10..10 or registrar dns:csps..323 gateway Media Termination Point (MTP) requirements: • If the Cisco Unified Border Element does H.323 calls. dtmf−relay h245−alphanumeric codec g711ulaw ! dial−peer voice 2 voip destination−pattern 8.323 or SIP unified communications trunk.323 Fast Start.323 Fast Start requirements: • If the Cisco Unified Border Element does H.4(6)T or later and the Cisco Unified Communications Manager is Version 4.150 incoming called−number 8. dial−peer voice 1 voip session target ipv4:10. .323 Fast Start. This implies the H.8. • An hardware or software MTP can be co−resident on the same router as the Cisco Unified Border Element (on routers platforms that support CUCM MTPs. H. an MTP is not mandatory as long as the Cisco Unified Border Element release is 12. Hence.com Interconnecting with Cisco Unified Communications Manager Cisco Unified Communications Manager can be interconnected with the Cisco Unified Border Element with the use of an H.information.13.1.16 dtmf−relay rtp−nte codec g711ulaw SIP User Agent Configuration Configure the SIP User Agent (UA) for registration and authentication.323 to SIP interworking for Cisco Unified Communications Manager.8.cisco. most SIP proxy servers require the SIP call to be Early Offer.225 trunk (GK controlled) toward Cisco Unified Border Element • Without a gatekeeperConfigure the Cisco Unified Border Element as an H.1. session protocol sipv2 session target ipv4:10.323 side must be H.

323 gateway on Cisco Unified Communications Manager. Configuration on the Cisco Unified Border Element for an H. Figure 1. Figure 2. The Configuration of the Cisco Unified Border Element as an H.323 Trunk .Figure 1 shows the configuration for a Cisco Unified Border Element defined as an H.323 Gateway on Cisco Unified Communications Manager Figure 2 shows the Cisco Unified Border Element configuration to match the preceding Cisco Unified Communications Manager configuration.

x or later is required to define a unified communications SIP trunk to the Cisco Unified Border Element. MTP requirements: • SIP trunk without an MTPConfigure a unified communications SIP trunk without MTP if delayed media or invite with no SDP is acceptable. Figure 3.711 calls only). • SIP trunk with MTPConfigure a unified communication SIP trunk (with MTP) if early media or invite with SDP is a requirement (G.SIP Trunk to the Cisco Unified Border Element Cisco Unified Communications Manager Version 5. Figures 3 shows the configuration for a Cisco Unified Border Element defined with a unified communications SIP trunk to Cisco Unified Communications Manager. Figure 4. The Configuration of the Cisco Unified Border Element with a SIP Trunk to Cisco Unified Communications Manager Figure 4 shows configuration the Cisco Unified Border Element configuration to match the preceding Cisco Unified Communications Manager configuration. Configuration on the Cisco Unified Border Element for a SIP Trunk .

Transcoding can be invoked for any call whether it originates from Cisco Unified Communications Manager towards the PSTN. or from the PSTN towards Cisco Unified Communications Manager.34.729.729. This is the configuration on the Cisco Unified Border Element for transcoding: voice−card 2 dspfarm dsp services dspfarm .711µ−law/a−law and various flavors of G.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register MTP ! dspfarm profile 1 mtp codec g711ulaw maximum sessions software 100 associate application SCCP Transcoding on the Cisco Unified Border Element The Cisco Unified Border Element can do transcoding between G. The configuration of transcoding on the Cisco Unified Border Element requires DSPs to be available on the platform.1 identifier 1 version 4. The main criterion is if the two call legs on the Cisco Unified Border Element have different codecs − G.MTP Co−resident with the Cisco Unified Border Element If a software MTP is required by the Cisco Unified Communications Manager configuration.711 and G.5. This is the configuration on the Cisco Unified Border Element for an MTP: sccp local FastEthernet0/1 sccp ccm 15. this can be configured on the same router used for the Cisco Unified Border Element.

1. There are a number of Tcl applications already built into Cisco IOS Software that can be used for Cisco Unified Border Element .100 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register MTP123456782012 keepalive retries 5 switchover method immediate switchback method immediate switchback interval 15 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec gsmfr codec g729r8 maximum sessions 5 associate application SCCP telephony−service load 7960−7940 P00303020214 max−ephones 48 max−dn 48 ip source−address 200.100 port 2000 sdspfarm units 1 sdspfarm transcode sessions 50 sdspfarm tag 1 MTP123456782012 create cnf−files version−stamp 7960 Jul 29 2002 13:50:03 Using Tcl IVR on the Cisco Unified Border Element The Cisco Unified Border Element supports Tcl scripts.1.1.1.sccp local FastEthernet 0/0 sccp ccm 200. and you can configure them under the VoIP dial−peers. There is no need for a DSP in order to use the Tcl functionality.

authorization. The Cisco IOS authentication.11/app_debitcard.5.11 auth−port 1645 acct−port 1646 radius−server timeout 10 radius−server key lab radius−server vsa send accounting radius−server vsa send authentication Full Sample Configuration router#show run Building configuration. and accounting (AAA) functionality can also be used in conjunction with Tcl scripting and the Cisco Unified Border Element to provide authentication and authorization of calls.3 service timestamps debug datetime msec service timestamps log datetime msec no service password−encryption ! hostname IPIPGW−1 ! boot−start−marker boot−end−marker ! .2.. aaa new−model ! aaa authentication login h323 group radius aaa authorization exec h323 local group radius aaa accounting exec h323 start−stop group radius ! application service debitcard tftp://15..27.27.8.5.5.27.2.deployments.0.tcl paramspace english index 1 paramspace english language en paramspace english location tftp://15.11/prompts/en/ param pid−len 4 paramspace english prefix en param uid−len 6 ! gw−accounting aaa ! radius−server host 15. Current configuration : 1122 bytes ! version 12.

no network−clock−participate aim 0 no network−clock−participate aim 1 no aaa new−model ip subnet−zero ip cef ! ! aaa new−model ! aaa authentication login h323 group radius aaa authorization exec h323 local group radius aaa accounting exec h323 start−stop group radius ! application service debitcard tftp://15.255.2.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ip classless ip route 0.0.1 ip http server ! control−plane .100 255.27.0.0 200.5.1.11/app_debitcard.255.27.0.8.11/prompts/en/ param pid−len 4 paramspace english prefix en param uid−len 6 ! gw−accounting aaa ! radius−server host 15.27.0.tcl paramspace english index 1 paramspace english language en paramspace english location tftp://15.1.5.1.2.1.11 auth−port 1645 acct−port 1646 radius−server timeout 10 radius−server key lab radius−server vsa send accounting radius−server vsa send authentication ! no ip domain lookup no ftp−server write−enable ! voice service voip allow−connections h323 to sip !−−− key command allow−connections sip to h323 !−−− key command allow−connections sip to sip !−−− key command allow−connections h323 to h323 !−−− key command ! interface FastEthernet0/0 ip address 200.0.5.0 0.

13.16 dtmf−relay rtp−nte !−−− DTMF config for RFC2833 codec g711ulaw ! gatekeeper shutdown sip−ua registrar ipv4:200..com authentication username xyz password xyz realm cisco.! dial−peer voice 1 voip application debitcard !−−− TCL Application session target ipv4:9. session protocol sipv2 session target ipv4:9. • loggingIt is important to ensure the Cisco Unified Border Element is set up for logging as in this example and also to perform debugging during non−peak hours as far as possible since the debug commands are verbose. dtmf−relay h245−alphanumeric !−−− DTMF config for h. logging logging service service console informational buffer 200000 debug sequence−number timestamp debug date msec .cisco.1. Note: Refer to Important Information on Debug Commands before you use debug commands.13.1. Troubleshooting Commands The Output Interpreter Tool ( registered customers only) (OIT) supports certain show commands.150 incoming called−number 8.245 alphanumeric codec g711ulaw ! dial−peer voice 2 voip destination−pattern 8..8.com ! line con 0 line aux 0 line vty 0 4 login ! end Verify There is currently no verification procedure available for this configuration. Use the OIT to view an analysis of show command output.8. Troubleshoot This section provides information you can use to troubleshoot your configuration...10 or registrar dns:csps.

2007 Document ID: 99863 . All rights reserved.323 to H.323 Scenarios debug debug debug debug debug debug debug debug debug h225 asn1 h225 q931 h225 events h245 asn1 h245 events h225 q931 cch323 all voip ipipgw voip ccapi inout H. Inc. Related Information • Voice Technology Support • Voice and Unified Communications Product Support • Recommended Reading: Troubleshooting Cisco IP Telephony • Technical Support & Documentation − Cisco Systems Contacts & Feedback | Help | Site Map © 2009 − 2010 Cisco Systems. these transcoder debugging commands should be enabled: debug dspfarm all debug sccp messages • debug voip rtp session named−eventsIf RFC2833 (dtmf−relay rtp−nte) is used. and get the output of the show logging command after the call has executed. H. you should also turn on this debug command. Terms & Conditions | Privacy Statement | Cookie Policy | Trademarks of Cisco Systems.• showThis is relevant output: show show show show version run voip rtp connection (once the call is up) call active voice brief (once the call is up) • debugMake sure to clear the log before a call for debugging is made.323 to SIP Scenarios debug debug debug debug debug debug debug debug debug h225 asn1 h225 q931 h225 events h245 asn1 h245 events cch323 all voip ipipgw voip ccapi inout ccsip all SIP to SIP Scenarios debug ccsip all debug voip ccapi inout • debugIn addition to the debugging commands based on the scenario described earlier. Inc. Updated: Oct 31.

Sign up to vote on this title
UsefulNot useful