- Cisco Unified Border Element Protocol-Independent Features and Setup
- Interworking Between RSVP Capable and RSVP Incapable Networks
- SIP INFO Method for DTMF Tone Generation
- WebEx Telepresence Media Support Over Single SIP Session
- DTMF Events through SIP Signaling
- Call Progress Analysis Over IP-to-IP Media Session
- Codec Preference Lists
- AAC-LD MP4A-LATM Codec Support on Cisco UBE
- Multicast Music-on-Hold Support on Cisco UBE
- Network-Based Recording
- Video Recording - Additional Configurations
- TDoS Attack Mitigation
- Cisco Unified Communications Gateway Services--Extended Media Forking
- Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls
- Acoustic Shock Protection
- Noise Reduction
- iLBC Support for SIP and H.323
- Configuring RTP Media Loopback for SIP Calls
- SIP Ability to Send a SIP Registration Message on a Border Element
- Session Refresh with Reinvites
- SIP Stack Portability
- Interworking of Secure RTP calls for SIP and H.323
- Cisco UBE Support for SRTP-RTP Internetworking
- Support for SRTP Termination
- Configuring RTCP Report Generation
- SIP SRTP Fallback to Nonsecure RTP
- Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks
- VoIP for IPv6
- Mid-call Signaling Consumption
- Support for Software Media Termination Point
- Cisco Unified Communication Trusted Firewall Control
- Cisco Unified Communication Trusted Firewall Control-Version II
- Domain-Based Routing Support on the Cisco UBE
- URI-Based Dialing Enhancements
- Fax Detection for SIP Call and Transfer