Table Of Contents

Feature Information for Cisco UBE Protocol-Independent Features and Setup

Feature Information for Cisco UBE Protocol-Independent Features and Setup

Table 1 lists the release history for this chapter.

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

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Note Table 1 lists only the Cisco IOS software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

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Table 1 Feature Information for CUBE Protocol-Independent Features and Setup Features 

Feature Name	Releases	Feature Information
Cisco Fax Relay	12.2(13)T	Fax relay is the default mode for passing faxes through a VoIP network, and Cisco fax relay is the default fax relay type on Cisco voice gateways.
Cisco IOS Tcl IVR and VoiceXML Application Guide	12.3(4)T	Tcl and VoiceXML applications on the Cisco gateway provide Interactive Voice Response (IVR) features and call control functionality such as call forwarding, conference calling, and voice mail.
Cisco Unified Border Element with Gatekeeper	12.4(4)T	Cisco Unified Border Element with Gatekeeper is designed to meet the interconnection needs of Internet telephony service providers (ITSPs) and of enterprises. One set of images provides basic interconnection and a second set provides interconnection through an Open Settlement Protocol (OSP) provider, enabling ITSPs to gain the benefits of the Cisco Unified Border Element with Gatekeeper while making use of the routing, billing, and settlement capabilities offered by OSP-based clearinghouses
Cisco Unified Communications Trusted Firewall	12.4(22)T	Cisco Unified Communications Trusted Firewall Control pushes intelligent services onto the network through a Trusted Relay Point (TRP) firewall. Firewall traversal is accomplished using Session Traversal Utilities for NAT(STUN) on a TRP colocated with a Cisco Unified Communications Manager Express (Cisco Unified CME) or a Cisco Unified Border Element.
Cisco Unified SIP Survivable Remote Site Telephony (SRST)	12.3(4)T	Cisco Unified SIP SRST provides backup to an external SIP proxy server by providing basic registrar and redirect server or back-to-back user agent (B2BUA) services.
Cisco VoiceXML Programmer's Guide	12.4(15)T	Voice Extensible Markup Language (VoiceXML) applications provide access to content and services over the telephone, just as Hypertext Markup Language (HTML) web pages provide access over a web browser residing on a PC.
Configuring Tool Command Language (Tcl)	—	The Tool Command Language (TCL) Interactive Voice Response (IVR) application programming interface (API) provides commands that you can use to write TCL scripts to interact with the Cisco IVR feature.
DTMF Events through SIP Signaling	12.2(11)T 12.2(8)YN 12.2(15)T 12.2(11)YV 12.2(11)T,	The DTMF Events through SIP Signaling feature provides the following: •DTMF event notification for SIP messages. •Capability of receiving hookflash event notification through the SIP NOTIFY method. •Third-party call control, or other signaling mechanisms, to provide enhanced services, such as calling card and messaging services. •Communication with the application outside of the media connection. The following commands were introduced or modified: timers notify and retry notify.
Dynamic payload type interworking for DTMF and codec packets for SIP-to-SIP calls	15.0(1)XA 15.1(1)T	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. The following commands were introduced or modified: asymmetric payload and voice-class sip asymmetric payload.
ENUM Support	12.4(6)T	The SIP-to-SIP Extended Feature Functionality Feature includes: •ENUM Support
H.323 RFC2833 - SIP NOTIFY	12.2(11)T	The SIP event notification mechanism uses NOTIFY messages to signal when certain telephony events take place. In order to send DTMF signals through NOTIFY messages, the gateway notifies the subscriber when DTMF digits are signaled by the originator. The notification contains a message body with a SIP response status line. This feature is introduced as part of the DTMF Events Through SIP Signaling feature set.
iLBC Support for SIP and H.323	12.2(11)T 12.2(15)T	The iLBC is a standard, high-complexity speech codec suitable for robust voice communication over IP. The iLBC has built-in error correction functionality that helps the codec perform in networks with high-packet loss. This codec is supported on both Session Initiation Protocol (SIP) and H.323. The following commands were introduced or modified: codec ilbc, codec preference, and rtp payload-type.
Interconnect RSVP capable and RSVP incapable networks	15.0(1)XA 15.1(1)T	Support for interworking between RSVP and non-RSVP call legs for SIP calls. This support includes •Early Offer to Early Offer calls •Delayed Offer to Delayed Offer calls •Delayed Offer to Early Offer calls Support for interworking between a non-RSVP H.323 call leg and RSVP SIP call leg include: •Fast Start to Early Offer calls •Slow Start to Delayed Offer calls
Interworking of Secure RTP calls for SIP and H.323	12.4(20)T	This feature provides an option for a Secure RTP (SRTP) call to be connected from H.323 to SIP and from SIP to SIP. Additionally, this feature extends SRTP fallback support from the Cisco IOS voice gateway to the Cisco Unified Border Element. This feature uses no new or modified commands.
Media Termination Point (MTP)	12.4(15)XY 15.0(1)M	Software Media Termination Point (MTP) provides the capability for Cisco Unified Communications Manager (Cisco UCM) to interact with a voice gateway via Skinny Client Control Protocol (SCCP) commands. These commands allow the Cisco UCM to establish an MTP for call signaling.
Negotiation of an Audio Codec from a List of Codecs on Each Leg of a SIP-to-SIP Call on the Cisco Unified Border Element	15.1(2)T	The Support for Negotiation of an Audio Codec from a List of Codecs on Each Leg of a SIP-to-SIP Call on the Cisco Unified Border Element feature supports negotiation of an audio codec using the Voice Class Codec and Codec Transparent infrastructure on the Cisco UBE. The following command was introduced or modified: voice-class codec (dial peer).
RSVP Agent	12.4(6)T	The RSVP Agent feature implements a Resource Reservation Protocol enables Cisco Unified Communications Manager to provide resource reservation for voice and video media to ensure QoS and call admission control (CAC).
RTP Media Loopback for SIP Calls	15.1(4)M	RTP packets are looped back toward the source when the RTP Media Loopback for SIP Calls feature is configured on a dial peer. SIP RTP media loopback helps in verifying the media path between the device originating the call and the intermediate device. The following commands were introduced or modified: None.
SIP DTMF Features	12.2(8)T 12.2(11)T	Provides support for dual-tone multifrequency (DTMF) signaling features: •RFC 2833 Dual-Tone Multifrequency (DTMF) Media Termination Point (MTP) Passthrough •DTMF Events Through SIP Signaling •DTMF Relay for SIP Calls Using Named Telephone Events •SIP INFO Method for DTMF Tone Generation •SIP NOTIFY-Based Out-of-Band DTMF Relay Support •SIP KPML-Based Out-of-Band DTMF Relay Support •SIP Support for Asymmetric SDP
SIP Parameter Modification	12.4(15)XZ 12.4(20)T	Allows users to change the standard SIP messages sent from the Cisco SIP stack for better interworking with different SIP entities. This feature introduces or modifies the following commands: voice class sip-profiles, voice-class sip profiles
SIP SRTP Fallback to Nonsecure RTP	12.4(22)T	The SIP SRTP Fallback to Nonsecure RTP feature enables a Cisco IOS Session Initiation Protocol (SIP) gateway to fall back from SRTP to RTP by accepting or sending an RTP/AVP(RTP) profile in response to an RTP/SAVP(SRTP) profile. This feature also allows inbound and outbound SRTP calls with nonsecure SIP signaling schemes (such as SIP URL) and provides the administrator the flexibility to configure TLS, IPsec, or any other security mechanism used in the lower layers for secure signaling of crypto attributes. The following commands were introduced or modified: srtp (voice), srtp negotiate, and voice-class sip srtp negotiate
SIP Video Calls with Flow Around Media	12.4(15)XZ 12.4(20)T	This feature provides the capability for media packets to pass directly between endpoints without the intervention of the Cisco UBE. The following command was modified by this feature: media
SIP Video Support for Telepresence Calls	—	This feature allows the Cisco Unified Border Element (Enterprise) to generate SIP INVITES that include SDP lines for both Voice and Voice media paths.
SIP—Ability to Send a SIP Registration Message on a Border Element	12.4(24)T	Provides the ability to send a SIP Registration Message from Cisco Unified Border Element. The following command was modified: credentials (SIP UA)
SIP—INFO Method for DTMF Tone Generation	12.2(11)T 12.3(2)T 12.2(8)YN 12.2(11)YV 12.2(11)T 12.2(15)T	The SIP—INFO Method for DTMF Tone Generation feature uses the Session Initiation Protocol (SIP) INFO method to generate dual-tone multifrequency (DTMF) tones on the telephony call leg. SIP methods, or request message types, request a specific action be taken by another user agent (UA) or proxy server. The SIP INFO message is sent along the signaling path of the call. The following command was introduced: show sip-ua.
SIP—SIP Stack Portability	12.4(2)T	Implements capabilities to the SIP gateway Cisco IOS stack involving user-agent handling of messages, handling of unsolicited messages, support for outbound delayed media, and SIP headers and content in requests and responses The following commands were introduced or modified: None
SIP-to-SIP Extended Feature Functionality for Session Border Controllers	12.4(6)T	The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-to-SIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs). The SIP-to-SIP Extended Feature Functionality includes: •Call Admission Control (based on CPU, memory, and total calls) •Delayed Media Call •ENUM Support •Configuring SIP Error Message Pass Through •Interoperability with Cisco Unified Communications Manager 5.0 and BroadSoft •Lawful Intercept •Media Inactivity •Modem Passthrough •TCP and UDP interworking •Tcl scripts with SIP NOTIFY VoiceXML with SIP-to-SIP •·Transport Layer Security (TLS)
Support for Interworking Between RSVP Capable and RSVP Incapable Networks	15.0(1)XA 15.1(1)T	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. The following commands were introduced or modified: acc-qos, ip qos defending-priority, ip qos dscp, ip qos policy-locator, ip qos preemption-priority, req-qos, voice-class sip rsvp-fail-policy,
T.38 Fax Relay	12.1(3)X1	This chapter describes how to configure T.38 fax relay on an IP network. It includes the following features: •Fax Relay Packet Loss Concealment •MGCP Based Fax (T.38) and DTMF Relay •SIP T.38 Fax Relay •T.38 Fax Relay for T.37/T.38 Fax Gateway •T.38 Fax Relay for VoIP H.323
Toll Fraud Prevention	—	—
Universal Transcoding	12.4(15)T	Universal Transcoding allows transcoding from any suppoted codec to any other supported codec.
VoIP Call Admission Control Using RSVP	12.1(5)T 12.2(11)T	Synchronizes RSVP signaling with H.323 Version 2 signaling to ensure that the bandwidth reservation is established in both directions before a call moves to the alerting phase (ringing). This ensures that the called party phone rings only after the resources for the call have been reserved. Using RSVP-based admission control, VoIP applications can reserve network bandwidth and react appropriately if bandwidth reservation fails.
VoIP Call Admissions Control	—	Call Admission Control (CAC) is a deterministic and informed decision that is made before a voice call is established and is based on whether the required network resources are available to provide suitable QoS for the new call.
VoIP for IPv6	12.4(22)T	VoIP for IPv6 •IPv4 to IPv6 Calls (SIP and SIP) •IPv6 to IPv6 Calls (SIP and SIP) •Support for Dual Stack ANAT

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