- Cisco Unified Border Element SIP Support
- SIP Core SIP Technology Enhancements
- Reporting End-of-Call Statistics in SIP BYE Message
- Configurable Hostname in Locally Generated SIP Headers
- SIP Parameter Modification
- SIP Session Timer Support
- SIP-to-SIP Basic Functionality for Session Border Controller
- SIP-to-SIP Supplementary Services for Session Border Controller
- Session Refresh with Reinvites
- Configuring Cisco UBE Out-of-dialog OPTIONS Ping for Specified SIP Servers or Endpoints
- Configuring an Error Response Code upon an Out-of-Dialog OPTIONS Ping Failure
- SIP INFO Method for DTMF Tone Generation
- SIP Enhanced 180 Provisional Response Handling
- Configuring SIP 181 Call is Being Forwarded Message
- Expires Timer Reset on Receiving or Sending SIP 183 Message
- SIP UPDATE Message per RFC 3311
- Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Configurable Pass-through of SIP INVITE Parameters
- Conditional Header Manipulation of SIP Headers
- Transparent Tunneling of QSIG and Q.931
- SIP Diversion Header Enhancements
- SIP History INFO
- SIP Ability to Send a SIP Registration Message on a Border Element
- Configurable SIP Parameters via DHCP
- Multiple Registrars on SIP Trunks
- Additional References
- Glossary
SIP-to-SIP Basic Functionality for Session Border Controller
The SIP-to-SIP Basic Functionality for Session Border Controller (SBC) for Cisco Unified Border Element (Cisco UBE) feature provides termination and re-origination of both signaling and media between VoIP and video networks using SIP signaling in conformance with RFC 3261. The SIP-to-SIP protocol interworking capabilities of the Cisco UBE support the following:
- Basic voice calls (Supported audio codecs include: G.711, G.729, G.728, G.726, G.723, G.722, AAC_LD, iLBC. Video codecs: H.263, and H.264)
- Codec transcoding
- Calling/called name and number
- Dual-Tone Multifrequency (DTMF) relay interworking
- SIP RFC 2833 <-> SIP RFC 2833
- SIP Notify <-> SIP Notify
- Interworking between SIP early-media and SIP early-media signaling
- Interworking between SIP delayed-media and SIP delayed-media signaling
- RADIUS call-accounting records
- Resource Reservation Protocol (RSVP) synchronized with call signaling
- SIP-SIP Video calls
- Tool Command Language Interactive Voice Response (TCL IVR) 2.0 for SIP, including media playout and digit collection (RFC 2833 DTMF relay)
- T.38 fax relay and Cisco fax relay
- UDP and TCP transport
Finding Feature Information
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 Feature Information Table at the end of this document.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Prerequisites for SIP-to-SIP Basic Functionality for Session Border Controller
Cisco Unified Border Element
- Cisco IOS Release 12.4(4)T or a later release must be installed and running on your Cisco Unified Border Element.
Cisco Unified Border Element (Enterprise)
- Cisco IOS XE Release 3.1S or a later release must be installed and running on your Cisco ASR 1000 Series Router.
Feature Information for SIP-to-SIP Basic Functionality for Session Border Controller
The following table provides release information about the feature or features described in this module. This table lists only the 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.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
ISR Feature History Information
| Table 1 | Feature Information for Configuring SIP-to-SIP Supplementary Features |
| Feature Name |
Releases |
Feature Information |
|---|---|---|
| SIP-to-SIP Basic Functionality for Session Border Controller |
12.4(4)T |
The SIP-to-SIP Basic Functionality for Session Border Controller (SBC) for Cisco Unified Border Element (Cisco UBE) feature provides termination and re-origination of both signaling and media between VoIP and video networks using SIP signaling in conformance with RFC 3261. This feature uses no new or modified commands. |
ASR Feature History Information
| Table 2 | Feature Information for Configuring SIP-to-SIP Supplementary Features |
| Feature Name |
Releases |
Feature Information |
|---|---|---|
| SIP-to-SIP Basic Functionality for Session Border Controller |
Cisco IOS XE Release 3.1S, Cisco IOS XE Release 3.3S |
The SIP-to-SIP Basic Functionality for Session Border Controller (SBC) for Cisco Unified Border Element (Cisco UBE) feature provides termination and re-origination of both signaling and media between VoIP and video networks using SIP signaling in conformance with RFC 3261. This feature uses no new or modified commands. |
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Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.
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