Support for SRTP Termination

This Support for SRTP Termination feature enables Cisco Unified Border Element (Cisco UBE) support for Secure Real-time Transport Protocol (SRTP) on the Session Initiation Protocol (SIP) Trunk interface.

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.

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.

Information About Support for SRTP Termination

The Support for SRTP Termination feature configures Cisco Unified Border Element (Cisco UBE) support for an Secure Real-time Transport Protocol (SRTP) connection using the AES_CM_128_HMAC_SHA1_80 crypto suite. This feature implements crypto-suite negotiation and appropriately sets up the call on the following two sides:
  • The Cisco Unified Call Manager (CUCM) or IP phones side—Connection between the end devices and CUBE
  • SIP Trunk side—Connection between CUBE and Service Provider

Prior to the Support for SRTP Termination feature, Cisco UBE could support an SRTP connection using the AES_CM_128_HMAC_SHA1_32 crypto suite. This crypto suite is still used by default, unless Cisco UBE is configured to use AES_CM_128_HMAC_SHA1_80 crypto suite.

Cisco UBE SRTP termination can be implemented in the following ways:

  • SRTP-RTP interworking—This method is used with devices (CUCM or IP Phone devices) that still support AES_CM_128_HMAC_SHA1_32 crypto suite only.

  • SRTP-SRTP pass-through—This method is used with devices that support AES_CM_128_HMAC_SHA1_80 crypto suite.

    Note
    This method of implementation is currently supported by non-CUCM end devices like Microsoft Link. This method can also be used when CUCM or IP phone devices support AES_CM_128_HMAC_SHA1_80 crypto suite.

For End Devices Supporting AES_CM_128_HMAC_SHA1_80 Crypto Suite

This method is used between Cisco Unified Border Element (Cisco UBE), IP Phones, and other Cisco Unified Call Manager (CUCM ) devices that support AES_CM_128_HMAC_SHA1_80 crypto suite.

  • CUCM or IP Phones side—A Secure Real-time Transport Protocol (SRTP) connection using the AES_CM_128_HMAC_SHA1_80 crypto suite exists here. In the figure below, IP Phone and CUBE within the customer network connect with an SRTP connection using AES_CM_128_HMAC_SHA1_80 crypto suite.

  • Session Initiation Protocol (SIP) Trunk side—An SRTP connection using the AES_CM_128_HMAC_SHA1_80 crypto suite. In the figure below, CUBE on the Customer Network and SBC on the Service Provider Network connect with an SRTP connection using the AES_CM_128_HMAC_SHA1_80 crypto suite.

Figure 1. SRTP Connection Supporting AES_CM_128_HMAC_SHA1_80 crypto suite



For End Devices Supporting AES_CM_128_HMAC_SHA1_32 Crypto Suite

A single Cisco Unified Call Manager (Cisco UBE) device cannot terminate a Secure Real-time Transport Protocol (SRTP) connection with an IP Phone using the AES_CM_128_HMAC_SHA1_32 crypto suite and initiate an SRTP connection with an external Cisco UBE device with the AES_CM_128_HMAC_SHA1_80 crypto suite at the same time.

For Cisco Unified Call Manager (CUCM) and IP Phone devices that support only AES_CM_128_HMAC_SHA1_32 crypto suite, the interim SRTP-RTP interworking solution that is described below can be implemented.

  • CUCM or IP Phone side:

    • An SRTP connection using the AES_CM_128_HMAC_SHA1_32 crypto suite exists between the IP Phone and CUBE1.

    • An RTP connection exists between CUBE1 and CUBE2.

  • SIP trunk side—An SRTP connection using the AES_CM_128_HMAC_SHA1_80 crypto suite is initiated by CUBE2 here. In the image below, CUBE2 is the border element on the Customer Network and SBC is the border element on the Service Provider Network.

Figure 2. SRTP-RTP Interworking Supporting AES_CM_128_HMAC_SHA1_32 crypto suite



How to Configure Support for SRTP Termination

Configuring Crypto Authentication

Configuring Crypto Authentication (Global Level)

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    voice service voip

    4.    sip

    5.    srtp-auth {sha1-32 | sha1-80}

    6.    end


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example: 
    Device> enable
     
    Enables privileged EXEC mode.
    • Enter your password if prompted.
     
    Step 2configure terminal


    Example: 
    Device# configure terminal
     

    Enters global configuration mode.

     
    Step 3voice service voip


    Example: 
    Device(config)# voice service voip
     

    Specifies VoIP encapsulation and enters voice-service configuration mode.

     
    Step 4sip


    Example: 
    Device(conf-voi-serv)# sip
     

    Enters the Session Initiation Protocol (SIP) configuration mode.

     
    Step 5srtp-auth {sha1-32 | sha1-80}


    Example: 
    Device(conf-serv-sip)# srtp-auth sha1-80
     
    Configures an SRTP connection on CUBE using the preferred crypto suite.

    • The default value is sha1-32.

     
    Step 6end


    Example: 
    Router(conf-serv-sip)# end
     

    Ends the current configuration session and returns to privileged EXEC mode.

     

    Configuring Crypto Authentication (Dial Peer Level)

    SUMMARY STEPS

      1.    enable

      2.    configure terminal

      3.    dial-peer voice tag voip

      4.    voice-class sip srtp-auth {sha1-32 | sha1-80 | system}

      5.    end


    DETAILED STEPS
       Command or ActionPurpose
      Step 1 enable


      Example: 
      Device> enable
       
      Enables privileged EXEC mode.
      • Enter your password if prompted.
       
      Step 2configure terminal


      Example: 
      Device# configure terminal
       

      Enters global configuration mode.

       
      Step 3dial-peer voice tag voip


      Example: 
      Device(config)# dial-peer voice 15 voip
       

      Defines a VoIP dial peer and enters dial peer voice configuration mode.

       
      Step 4voice-class sip srtp-auth {sha1-32 | sha1-80 | system}


      Example: 
      Device(config-dial-peer)# voice-class sip srtp-auth sha1-80
       
      Configures an SRTP connection on CUBE using the preferred crypto suite.

      • The default value is sha1-32.

       
      Step 5end


      Example: 
      Router(conf-serv-sip)# end
       

      Ends the current configuration session and returns to privileged EXEC mode.

       

      Verifying Support for SRTP Termination

      Perform this task to verify the configuration of an SRTP connection on Cisco Unified Border Element using the AES_CM_128_HMAC_SHA1_80 crypto suite. The show commands can be entered in any order.

      SUMMARY STEPS

        1.    show sip-ua calls

        2.    show sip-ua srtp


      DETAILED STEPS
        Step 1   show sip-ua calls


        Example:

        The following example displays sample output for active user agent client (UAC) and user agent server (UAS) information on Session Initiation Protocol (SIP) calls: 

        Device# show sip-ua calls
Call 1
SIP Call ID                : 20894
   Media Stream 1
     Local Crypto Suite    : AES_CM_128_HMAC_SHA1_80
     Remote Crypto Suite: AES_CM_128_HMAC_SHA1_80 (AES_CM_128_HMAC_SHA1_80 AES_CM_128_HMAC_SHA1_32 )
        Step 2   show sip-ua srtp


        Example:

        The following example displays sample output for Session Initiation Protocol (SIP) user-agent (UA) SRTP information: 

        Device# show sip-ua srtp
SIP UA SRTP
Crypto-suite Negotiation
 AES_CM_128_HMAC_SHA1_80:  3
 AES_CM_128_HMAC_SHA1_32:  2

        Configuration Examples for Support for SRTP Termination

        Example: Configuring Crypto Authentication

        Example: Configuring Crypto Authentication (Global Level)

        The following example shows how to configure Cisco UBE to support an SRTP connection using the AES_CM_128_HMAC_SHA1_80 crypto suite at the global level: 

        Device> enable
Device# configure terminal
Device(config)# voice service voip
Device(conf-voi-serv)# sip
Device(conf-serv-sip)# srtp-auth sha1-80
Device(conf-serv-sip)# end

        Example: Configuring Crypto Authentication (Dial Peer Level)

        The following example shows how to configure Cisco UBE to support an SRTP connection using the AES_CM_128_HMAC_SHA1_80 crypto suite at the dial peer level: 

        Device> enable
Device# configure terminal
Device(config)# dial-peer voice 15 voip
Device(config-dial-peer)# voice-class sip srtp-auth sha1-80
Device(config-dial-peer)# end

        Additional References for Support for SRTP Termination

        Related Documents

        Related Topic Document Title
        Voice commands Cisco IOS Voice Command Reference
        Cisco IOS commands Cisco IOS Master Command List, All Releases
        SIP configuration tasks SIP Configuration Guide, Cisco IOS Release 15M&T

        Technical Assistance

        Description Link

        The Cisco Support website provides extensive online resources, including documentation and tools for troubleshooting and resolving technical issues with Cisco products and technologies.

        To receive security and technical information about your products, you can subscribe to various services, such as the Product Alert Tool (accessed from Field Notices), the Cisco Technical Services Newsletter, and Really Simple Syndication (RSS) Feeds.

        Access to most tools on the Cisco Support website requires a Cisco.com user ID and password.

        http:/​/​www.cisco.com/​support

        Feature Information for Support for SRTP Termination

        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 . An account on Cisco.com is not required.
        Table 1 Feature Information for Support for SRTP Termination

        Feature Name

        Releases

        Feature Information

        Support for SRTP Termination

        15.4(1)T

        The Support for SRTP Termination feature describes how to configure Cisco Unified Border Element to support AES_CM_128_HMAC_SHA1_80 crypto suite on the Session Initiation Protocol (SIP) Trunk interface.

        The following commands were introduced or modified: show sip-ua srtp, srtp-auth and voice-class sip srtp-auth.