Table Of Contents
Cisco Unified Border Element Features Roadmap
Cisco Unified Border Element Features Roadmap
Revised: October 27, 2009First Published: March 18, 2005Last Updated: October 27, 2009This roadmap lists the features documented in the Cisco Unified Border Element Configuration guide (previously known as the Cisco Multiservice IP-to-IP Gateway Application Guide) and maps them to the chapters in which they appear.
Activation Before you can configure the software features described in this guide, you will need a Product Authorization Key (PAK). Before you start the configuration process, please register your products and activate your PAK at the following URL http://www.cisco.com/go/license.Your software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the "Cisco Unified Border Element Features Roadmap" section.
Use Cisco Feature Navigator to find information about platform support and Cisco IOS and Catalyst OS software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration Library—including feature documents, and troubleshooting information—at http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124tcg/vcl.htm.
Note
Table 1 lists only the Cisco IOS software release that introduced support for a given feature in a given Cisco IOS software release train. Unless noted otherwise, subsequent releases of that Cisco IOS software release train also support that feature.
Table 1 Supported Cisco Unified Border Element Configuration Guide Features
Support for Interworking Between RSVP Capable and RSVP Incapable Networks
The Support for Interworking Between RSVP Capable and RSVP Incapable Networks feature provides precondition-based RSVP support for basic audio call and supplementary services on the Cisco UBE.
"Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks" in the "H.323-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP to SIP Calls
The Support for Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP to SIP Calls feature provides dynamic payload type interworking for DTMF and codec packets for SIP to SIP calls.
"Configuring Support for Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls Feature" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Error Response Code upon an Out-of-Dialog OPTIONS Ping Failure
This feature provides option to configure the error response code when a dial peer is busied out because of an Out-of-Dialog OPTIONS ping failure.
"Configuring an Error Response Code upon an Out-of-Dialog OPTIONS Ping Failure" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Cisco UBE MIB support
Supports CISCO-VOICE-DIAL-CONTROL-MIB objects to obtain call volume and call rate information and CISCO-DSP-MGMT-MIB objects to report transcoding sessions availability information on the Cisco Unified Border Element.
"Related Documents" section of the, "H.323-to-H.323 Connections on a Cisco Unified Border Element", "H.323-to-SIP Connections on a Cisco Unified Border Element" and the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapters of this guide
Adjustable Timers for Registration Refresh and Retries
This feature provides the ability for IOS software to:
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"Configuring Adjustable Timers for Registration Refresh and Retries" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Cisco Unified Border Element Support for SRTP-RTP Internetworking
This feature allows secure enterprise-to-enterprise calls. Support for SRTP-RTP internetworking between one or multiple Cisco Unified Border Elements is enabled for SIP-SIP audio calls.
"Cisco Unified Border Element Support for SRTP-RTP Internetworking" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Configurable SIP Parameters via DHCP
The Configurable SIP Parameters via DHCP feature introduces the configuring of SIP parameters via DHCP.
"Configurable SIP Parameters via DHCP" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Forced Update to SIP Parameters via DHCP updated in FTS.
The Configurable SIP Parameters via DHCP feature introduces the configuring of SIP parameters via DHCP.
"Enabling Forced Update of SIP Parameters via DHCP" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Out-of-dialog OPTIONS Ping to Monitor Dial-peers to Specified SIP Servers and Endpoints
This feature provides a keepalive mechanism at the SIP level between any number of destinations. The generic heartbeat mechanism allows Cisco UBE to monitor the status of SIP servers or endpoints and provide the option of busying-out associated dial-peer upon total heartbeat failure.
"Configuring Cisco UBE Out-of-dialog OPTIONS Ping for Specified SIP Servers or Endpoints" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Preloaded Routes in Outgoing INVITE Messages Based on REGISTER Information
This feature supports the use of preloaded routes for outgoing INVITE messages. The system routes INVITE messages based on REGISTER message information, such as the path: and Service-Route values
"Support for Preloaded Routes in Outgoing INVITE Messages Based on REGISTER Information"in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Selectively Using sip: URI or tel: URL Formats on Individual SIP Headers
This feature supports the construction of request URIs in tel: format. The system supports this format for both the To: header and the Request-Line. The system also supports appending the phone-context to the tel: URL.
"Selectively Using sip: URI or tel: URL Formats on Individual SIP Headers" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Support for PAID, PPID, Privacy, PCPID, and PAURI Headers on the Cisco UBE
This feature provides the ability to:
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"Support for PAID, PPID, Privacy, PCPID, and PAURI Headers on the Cisco Unified Border Element" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Unsupported Content Pass-through
The feature introduces the ability to configure the Cisco UBE to pass through end to end headers at a global or dial-peer level, that are not processed or understood in a SIP trunk to SIP trunk scenario. The pass through functionality includes all or only a configured list of unsupported or non-mandatory SIP headers, and all unsupported content/MIME types.
"Configuring Cisco Unified Border Element for Unsupported Content Pass-through" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
SIP Registration Message
Provides the ability to send a SIP Registration Message from Cisco Unified Border Element using the credentials command.
"Configuring Cisco Unified Border Element for Unsupported Content Pass-through" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Adjustable Timers for Registration Refresh and Retries
This feature provides the ability for IOS software to:
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"Configuring Adjustable Timers for Registration Refresh and Retries" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Cisco Unified Border Element Support for SRTP-RTP Internetworking
This feature allows secure enterprise-to-enterprise calls. Support for SRTP-RTP internetworking between one or multiple Cisco Unified Border Elements is enabled for SIP-SIP audio calls.
"Cisco Unified Border Element Support for SRTP-RTP Internetworking" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Configurable SIP Parameters via DHCP
The Configurable SIP Parameters via DHCP feature introduces the configuring of SIP parameters via DHCP.
"Configurable SIP Parameters via DHCP" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Forced Update to SIP Parameters via DHCP updated in FTS.
The Configurable SIP Parameters via DHCP feature introduces the configuring of SIP parameters via DHCP.
The following sections provide information about this feature:
"Enabling Forced Update of SIP Parameters via DHCP" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Out-of-dialog OPTIONS Ping to Monitor Dial-peers to Specified SIP Servers and Endpoints
This feature provides a keepalive mechanism at the SIP level between any number of destinations. The generic heartbeat mechanism allows Cisco UBE to monitor the status of SIP servers or endpoints and provide the option of busying-out associated dial-peer upon total heartbeat failure.
"Configuring Cisco UBE Out-of-dialog OPTIONS Ping for Specified SIP Servers or Endpoints" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Preloaded Routes in Outgoing INVITE Messages Based on REGISTER Information
This feature supports the use of preloaded routes for outgoing INVITE messages. The system routes INVITE messages based on REGISTER message information, such as the path: and Service-Route values
"Support for Preloaded Routes in Outgoing INVITE Messages Based on REGISTER Information"in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Selective Filtering of Outgoing Provisional Response on the Cisco Unified Border Element
Supports selective filtering of outgoing provisional responses, including 180 - Alerting, and 183-Session In Progress responses. Selective filtering can be further based on the availability of media information in the received provisional response.
"Configuring Selective Filtering of Outgoing Provisional Response on the Cisco Unified Border Element" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Selectively Using sip: URI or tel: URL Formats on Individual SIP Headers
This feature supports the construction of request URIs in tel: format. The system supports this format for both the To: header and the Request-Line. The system also supports appending the phone-context to the tel: URL.
"Selectively Using sip: URI or tel: URL Formats on Individual SIP Headers" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Support for PAID, PPID, Privacy, PCPID, and PAURI Headers on the Cisco UBE
This feature provides the ability to:
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"Support for PAID, PPID, Privacy, PCPID, and PAURI Headers on the Cisco Unified Border Element" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Unsupported Content Pass-through
The feature introduces the ability to configure the Cisco UBE to pass through end to end headers at a global or dial-peer level, that are not processed or understood in a SIP trunk to SIP trunk scenario. The pass through functionality includes all or only a configured list of unsupported or non-mandatory SIP headers, and all unsupported content/MIME types.
"Configuring Cisco Unified Border Element for Unsupported Content Pass-through" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide.
Delayed offer to Early offer for SIP Video Calls
Forces a Cisco Unified Border Element to send a SIP invite with Early-Offer (EO) on the Out-Leg (OL).
"Configuring Delayed-Offer to Early-Offer for SIP Video Calls" in the "Configuring Cisco Unified Border Element Videoconferencing" chapter of this guide
Configurable SIP Listening Port
Allows users the ability to configure the port that SIP messages are listened on.
"Configuring SIP Listening Port" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Configuring Bandwidth Parameters for SIP Calls
This features provides the ability to manually configure the bandwidth that is signaled in the outbound SIP invite.
"Configuring Bandwidth Parameters for SIP Calls" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Product Authorization Key (PAK)
Requires users to register products and activate a Product Authorization Key (PAK) before starting the configuration process.
Note
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H.323 Calling Name Display
Provides a configurable option on the Cisco Gateway to send and interpret the calling name information received in Q931 Facility messages so that the Cisco Unified Communications Manager can display the calling name on the Cisco IP Phones.
"Configuring H.323 Calling Name Display" in the "Cisco Unified Border Element for H.323 Cisco Unified Communications Manager to H.323 Service Provider Connectivity" chapter of this guide
Session Border Controller Enhancements for H.323-to-SIP and SIP-to-SIP Supplementary Services, Transcoding Optimization and Firewall Integration.
New H.323-to-SIP features offered in this release include:
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New SIP-to-SIP features offered in this release include:
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New features offered in this release include:
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"Overview of Cisco Unified Border Element", "H.323-to-SIP Connections on a Cisco Unified Border Element", and "SIP-to-SIP Connections on a Cisco Unified Border Element" chapters of this guide
Session Refresh with reinvites
This feature expands the ability of the Cisco Unified BE to receive a REINVITE that contains either a session refresh parameter or a change in media via a new SDP and ensure the session does not time out.
"Configuring Support for Session Refresh with Reinvites" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
SIP Video Calls with Flow Around Media
Allows SIP video calls where the media flows around the Cisco Unified Border Element.
"SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Configurable SIP Parameters
Allows users to change the standard SIP messages sent from the Cisco SIP stack for better interworking with different SIP entities.
"Configuring SIP Parameters" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Product Authorization Key (PAK)
Requires users to register products and activate a Product Authorization Key (PAK) before starting the configuration process.
Note
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Configurable SIP Listening Port
Allows users the ability to configure the port that SIP messages are listened on.
"Configuring SIP Listening Port" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Configuring Bandwidth Parameters for SIP Calls
This features provides the ability to manually configure the bandwidth that is signaled in the outbound SIP invite.
"Configuring Bandwidth Parameters for SIP Calls" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Session Refresh with reinvites
This feature expands the ability of the Cisco Unified BE to receive a REINVITE that contains either a session refresh parameter or a change in media via a new SDP and ensure the session does not time out.
"Configuring Support for Session Refresh with Reinvites" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Delayed offer to Early offer for SIP Video Calls
Forces a Cisco Unified Border Element to send a SIP invite with Early-Offer (EO) on the Out-Leg (OL).
"Configuring Delayed-Offer to Early-Offer for SIP Video Calls" in the "Configuring Cisco Unified Border Element Videoconferencing" chapter of this guide
Session Border Controller Enhancements for H.323-to-SIP and SIP-to-SIP Supplementary Services, Transcoding Optimization and Firewall Integration.
New H.323-to-SIP features offered in this release include:
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New SIP-to-SIP features offered in this release include:
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New features offered in this release include:
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"Overview of Cisco Unified Border Element", "H.323-to-SIP Connections on a Cisco Unified Border Element", and "SIP-to-SIP Connections on a Cisco Unified Border Element" chapters of this guide
Ability to Send a SIP Registration Message on a Cisco Unified Border Element
This feature introduces a new command that allows the Cisco Unified Border Element to send a REGISTRATION command to a SIP REGISTRAR.
"Configuring SIP Listening Port" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
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The official marketing name of Cisco Multiservice IP-to-IP Gateway was changed to Cisco Unified Border Element (Cisco UBE).
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Interworking of Secure RTP calls for SIP and H323
New features offered in this release include:
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H.323 Video Calls Support for H.239 Signalling
New features offered in this release include:
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H323 Video Calls Support for H.235 Security
New features offered in this release include:
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Delayed offer to Early offer for SIP Audio Calls
Forces a Cisco Unified Border Element to send a SIP invite with Early-Offer (EO) on the Out-Leg (OL).
"Configuring Delayed-Offer to Early-Offer for SIP Audio Calls" in the "SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Lawful Intercept for Cisco 3825
Adds Lawful Intercept support on the Cisco 3825.
"Overview of Cisco Unified Border Element" chapter of this guide.
H.323 Calling Name Display
Provides a configurable option on the Cisco Gateway to send and interpret the calling name information received in Q931 Facility messages so that the Cisco Unified Communications Manager can display the calling name on the Cisco IP Phones.
"Configuring H.323 Calling Name Display" in the "Cisco Unified Border Element for H.323 Cisco Unified Communications Manager to H.323 Service Provider Connectivity" chapter of this guide
Enhanced Hosted NAT Traversal and IP Call Leg Statistics for Session Border Controller (SBC)
Provides enhanced termination and re-origination of signaling and media between VoIP and Video Networks in conformance with RFC3261.
New features offered in this release include:
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New H.323-to-H.323 features offered in this release on the Cisco 28xx, 38xx, 5350XM and 5400XM include:
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New H.323-to-SIP features offered in this release on the Cisco 28xx, 38xx, 5350XM and 5400XM include:
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New SIP-to-SIP features offered in this release on the Cisco 28xx, 38xx, 5350XM and 5400XM include:
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"Overview of Cisco Unified Border Element", "H.323-to-H.323 Connections on a Cisco Unified Border Element"", "H.323-to-SIP Connections on a Cisco Unified Border Element", "SIP-to-SIP Connections on a Cisco Unified Border Element" chapters of this guide
H.323-to-SIP Supplementary Feature Interworking for Session Border Controller (SBC)
New H.323-to-H.323 features offered in this release include:
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New H.323-to-SIP features offered in this release include:
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New SIP-to-SIP features offered in this release include:
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"Fundamental Cisco Unified Border Element Configuration", "H.323-to-H.323 Connections on a Cisco Unified Border Element", "H.323-to-SIP Connections on a Cisco Unified Border Element", "SIP-to-SIP Connections on a Cisco Unified Border Element" chapters of this guide
DTMF Relay Digit-Drop on an Cisco Unified Border Element with Cisco Unified Communications Manager
This feature passes DTMF tones out-of-band and drops in-band digits to avoid sending both tones to the outgoing leg on an H.323-to-SIP Cisco UBE.
"Cisco Unified Border Element for H.323 Cisco Unified Communications Manager to H.323 Service Provider Connectivity" chapter of this guide.
H.323-to-SIP Supplementary Feature Interworking for Session Border Controller (SBC)
New H.323-to-SIP features available include:
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New SIP-to-SIP features offered in this release include:
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"H.323-to-SIP Connections on a Cisco Unified Border Element" and "SIP-to-SIP Connections on a Cisco Unified Border Element" chapters of this guide
GSMAMR-NB Codec support on a Cisco Unified Border Element
Support for the complexity multimode codec that supports eight narrowband speech encoding modes with bit rates between 4.75 and 12.2 kbps.
"Fundamental Cisco Unified Border Element Configuration", chapter of this guide
SIP-to-SIP Supplementary Services for Session Border Controller (SBC)
New SIP-to-SIP features available include:
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"SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
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The Gatekeeper content from the Cisco Multiservice IP-to-IP Gateway Application guide was moved to a separate book located at the following
http://cisco.com/en/US/docs/ios/voice/cubegk/configuration/guide/ve_book/ve_book.html
Cisco Unified Border Element with Gatekeeper Configuration Guide
SIP-to-SIP Extended Feature Functionality for Session Border Controller (SBC)
Enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP UAs. New SIP-to-SIP features available include:
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"SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
H.323-to-SIP Extended Call Interworking for Session Border Controller (SBC)
New H.323-to-SIP features available include:
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"H.323-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
H.323-to-H.323 Extended Call Interworking for Session Border Controller (SBC)
New H.323-to-H.323 features available include:
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"H.323-to-H.323 Connections on a Cisco Unified Border Element" chapter of this guide
Interoperability Enhancements to the Cisco Unified Border Element
Enables operation of Cisco Unified Border Element features concurrently on the same router with H.323 gatekeeper and TDM-IP voice-gateway features.
"Fundamental Cisco Unified Border Element Configuration"chapter of this guide
SIP-to-H.323 Extended Call Interworking for Session Border Controller (SBC)
Enables the Cisco UBE to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323).
"H.323-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
SIP-to-SIP Basic Functionality for Session Border Controller (SBC)
Enables the Cisco UBE to bridge calls between SIP networks.
"SIP-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Support for Cisco UBE and Gatekeeper Features on the Cisco 2801
Provides integrated voice and video services on the Cisco 2801.
"Fundamental Cisco Unified Border Element Configuration" chapter of this guide
Scalability Enhancements for the Cisco 7301 + 7200 NPE-G1 (Extended Socket Boundary)
Increases the call capacity of the Cisco UBE by extending the total number of sockets supported on the Cisco 7301 and Cisco 7200 NPE-G1 routers.
Integrated into Cisco IOS software. No configuration is required.
Support for Cisco Unified Border Element and Gatekeeper Features on the Cisco 2800 Series and Cisco 3800 Series
Provides integrated voice and video services on the Cisco 2800 series and Cisco 3800 series.
"Fundamental Cisco Unified Border Element Configuration" and chapter of this guide (see also Table 1 in the Overview of Cisco Unified Border Element chapter in this guide)
SIP-to-H.323 Basic Call Interworking for Session Border Controller (SBC)
Enables the Cisco Unified Border Element to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323).
"H.323-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
SIP-to-H.323 Dual Tone Multifrequency Relay Digit-Drop
Passes DTMF tones out-of-band and drops in-band digits to avoid sending both tones to the outgoing leg on H.323-to-SIP Cisco UBE.
"H.323-to-SIP Connections on a Cisco Unified Border Element" chapter of this guide
Call Failure Recovery (Rotary) on the Cisco Unified Border Element
Eliminates codec restrictions and enables the Cisco UBE to restart codec negotiation with the originating endpoint based on the codec capabilities of the next dial peer in the rotary group for H.323-to-H.323 interconnections.
"H.323-to-H.323 Connections on a Cisco Unified Border Element" chapter of this guide
H.323-to-H.323 Interworking Between FastStart and Normal H.245 Signaling
Enables the Cisco UBE to bridge calls between VoIP endpoints that support only H.323 FastStart procedures and endpoints that support only normal H.245 signaling (SlowStart).
"H.323-to-H.323 Connections on a Cisco Unified Border Element" chapter of this guide
Transcoding G.711-G.729
Supports transcoding (compression and decompression of voice streams to match endpoint-device capabilities) between G.711 and G.729 codecs when the router chassis is equipped with DSP resources (H.323-H.323 and H.323-SIP).
"Fundamental Cisco Unified Border Element Configuration" chapter of this guide
Interoperability Enhancements to the Cisco Unified Border Element
Enables operation of Cisco UBE features concurrently on the same router with H.323 gatekeeper and TDM-IP voice-gateway features. Supports interoperability with the Cisco ATA-188 and with Microsoft NetMeeting.
"Fundamental Cisco Unified Border Element Configuration" chapter of this guide
Videoconferencing for the Cisco Unified Border Element Feature
Adds video capabilities and improved QoS, allowing increased scalability and control for IP telephony and IP videoconferencing networks.
"Configuring Cisco Unified Border Element Videoconferencing" chapter of this guide
Cisco Unified Border Element for H.323 Cisco Unified Communications Manager to H.323 Service Provider Connectivity
Provides interoperability with Cisco Unified Communications Manager for basic calls, caller-ID services, supplementary services, and RSVP synchronization with audio.
"H.323-to-H.323 Connections on a Cisco Unified Border Element" chapter of this guide
Cisco Unified Border Element with Media Flow-Around
Adds media flow-around capability on the Cisco UBE by supporting the processing of call setup and teardown requests (VoIP call signaling) and for media streams (flow-through and flow-around). Improves scalability and performance when network-topology hiding and bearer-level interworking features are not required.
How to Configuring Media Flow-Around sections of the, "H.323-to-H.323 Connections on a Cisco Unified Border Element", "H.323-to-SIP Connections on a Cisco Unified Border Element", and "SIP-to-SIP Connections on a Cisco Unified Border Element" chapters of this guide.
H.323 Cisco Unified Border Element
Provides a network-to-network demarcation point between independent VoIP and video networks by for billing, security, call-admission control, QoS, and signaling interworking. Performs most of the functions of a PSTN-to-IP gateway but joins two H.323 VoIP call legs.
"Fundamental Cisco Unified Border Element Configuration" chapter of this guide
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