- Interworking Between RSVP Capable and RSVP Incapable Networks
- SIP INFO Method for DTMF Tone Generation
- DTMF Events through SIP Signaling
- Negotiation of an Audio Codec from a List of Codecs
- Multicast Music-on-Hold Support on Cisco UBE
- Network-Based Recording Using Cisco UBE
- Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls
- iLBC Support for SIP and H.323
- Configuring RTP Media Loopback for SIP Calls
- Support for Media Flow- Around with SIP Signaling control on CUBE
- Configuring Media Antitrombone
- SIP Ability to Send a SIP Registration Message on a Border Element
- SIP Parameter Modification
- Session Refresh with Reinvites
- SIP Stack Portability
- Interworking of Secure RTP calls for SIP and H.323
- CUBE Support for SRTP-RTP Internetworking
- Configuring RTCP Report Generation
- SIP SRTP Fallback to Nonsecure RTP
- Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks
- VoIP for IPv6
- Cisco UBE Mid-call Re-INVITE Consumption
- Interworking Between RSVP Capable and RSVP Incapable Networks
- 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
Cisco UBE Protocol-Independant Features and Setup Cisco IOS Release 15.2MT
Bias-Free Language
The documentation set for this product strives to use bias-free language. For the purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on RFP documentation, or language that is used by a referenced third-party product. Learn more about how Cisco is using Inclusive Language.
- Updated:
- August 3, 2011
Chapter: Interworking of Secure RTP calls for SIP and H.323
Interworking of Secure RTP calls for SIP and H.323
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 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 Interworking of Secure RTP calls for SIP and H.323
The following are prerequisites for the Interworking of Secure RTP calls for SIP and H.323 feature:
- Establish a working IP network and configure VoIP.
 Note |
For information about configuring VoIP, see Enhancements to the Session Initiation Protocol for VoIP on Cisco Access Platforms at the following URL: http://www.cisco.com/en/US/docs/ios/12_2t/12_2t11/feature/guide/ftsipgv1.html |
- Ensure that the gateway has voice functionality configured for SIP.
- Ensure that your Cisco router has adequate memory.
- As necessary, configure the router to use Greenwich Mean Time (GMT). SIP requires that all times be sent in GMT. SIP INVITE messages are sent in GMT. However, the default for routers is to use Coordinated Universal Time (UTC). To configure the router to use GMT, issue the clock timezone command in global configuration mode and specify GMT.
Cisco Unified Border Element
- Cisco IOS Release 12.2(20)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.
Restrictions for Interworking of Secure RTP calls for SIP and H.323
- The SIP gateway does not support codecs other than those listed in the table titled "SIP Codec Support by Platform and Cisco IOS Release" in the "Enhanced Codec Support for SIP Using Dynamic Payloads" section of the Configuring SIP QoS Features module at the following URL: http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-qos.html
- SIP requires that all times be sent in GMT.
Feature Information for Configuring Interworking of Secure RTP Calls for SIP and H.323
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.
Feature History Table entry for the Cisco Unified Border Element.
| Table 1 | Feature Information for Configuring Support for Expires Timer Reset on Receiving or Sending SIP 183 Message |
| Feature Name |
Releases |
Feature Information |
|---|---|---|
| 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. |
Feature History Table entry for the Cisco Unified Border Element (Enterprise).
| Table 2 | Feature Information for Configuring Support for Expires Timer Reset on Receiving or Sending SIP 183 Message |
| Feature Name |
Releases |
Feature Information |
|---|---|---|
| Interworking of Secure RTP calls for SIP and H.323 |
Cisco IOS XE Release 3.1S |
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. |
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