The Session Initiation Protocol (SIP) support for the Secure Real-time Transport Protocol (SRTP) is an extension of the Real-time Transport Protocol (RTP) Audio/Video Profile (AVP) and ensures the integrity of RTP and Real-Time Control Protocol (RTCP) packets that provide authentication, encryption, and the integrity of media packets between SIP endpoints.
SIP support for SRTP was introduced in Cisco IOS Release 12.4(15)T. In this and later releases, you can configure the handling of secure RTP calls on both a global level and on an individual dial peer basis on Cisco IOS voice gateways. You can also configure the gateway (or dial peer) either to fall back to (nonsecure) RTP or to reject (fail) the call for cases where an endpoint does not support SRTP.
The option to allow negotiation between SRTP and RTP endpoints was added for Cisco IOS Release 12.4(20)T and later releases, as was interoperability of SIP support for SRTP on Cisco IOS voice gateways with Cisco Unified Communications Manager. In Cisco IOS Release 12.4(22)T and later releases, you can also configure SIP support for SRTP on Cisco Unified Border Elements (Cisco UBEs).
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest caveats and feature information, see
Bug Search Tool and 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 module.
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.