Cisco Unified Border Element (Enterprise) SIP Support Configuration Guide, Cisco IOS XE Release 3S
Cisco UBE Mid-call Re-INVITE/UPDATE Consumption
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Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

Contents

Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

The Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature helps to avoid interoperability issues by consuming the mid-call Re-INVITEs/UPDATEs from Cisco Unified Border Element.

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.

Prerequisites for Cisco UBE Mid-call Re-INVITE Consumption

Cisco Unified Border Element
  • Cisco IOS Release 15.2(1)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 3.6S or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Restrictions for Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

  • The Cisco UBE Mid-call Re-INVITE consumption feature works only in Cisco UBE and Session Initiation Protocol - Session Initiation Protocol (SIP) calls and not for SIP-H.323 or in TDM Gateways.
  • Cisco UBE Session Description Protocol (SDP) -passthrough does not work with Cisco UBE Mid-call Re-INVITE consumption.
  • When codec T is configured, the offer from Cisco UBE will only have audio codecs, and not video codecs and so the video codecs are not consumed.
  • Cisco UBE Mid-call Re-INVITE/UPDATE consumption works only if the flow-throughcommand is configured. If the flow-around command is configured, Re-INVITEs/UPDATEs will not be consumed.
  • Media Antitrombones are media loops in a SIP entity due to call transfer or call forward. When Media Antitrombone feature is configured, Cisco UBE moves dynamically to flow-aroundas a result media antitrombone takes precedence over the Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature.
  • Video transcoding is not supported when Mid-call Re-INVITE consumption is configured.
  • Secure Real-time Protocol - Real-time Protocol (SRTP-RTP) supplementary services and mid-call signaling are configured, so the Re-INVITEs/UPDATEs are not consumed.
  • Multicast Music On Hold (MMOH) is not supported and the Re-INVITEs/UPDATEs are passed over.
  • When the midcall-signaling passthru media-change command is configured and high-density transcoder is enabled, there might be some impact on Digital Signal Processing (DSP) resources as the transcoder might be used for all the calls.
  • Session timer is handled leg by leg whenever this feature is configured.

Information About Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

To configure Cisco UBE Mid-call Re-INVITE you should understand the following concepts:

Types of Mid-call Re-INVITE/UPDATE Consumption

Mid-call Re-INVITEs/UPDATEs consumption can be done in the following two ways:

Passthrough Media Change

Passthrough media change method optimizes or consumes mid-call, media-related signaling within the call. Mid-call signaling changes will be passed through only when bidirectional media like T.38 or video is added. The command midcall-signaling passthru media-change needs to be configured to enable passthrough media change.

Block

The Block method blocks all mid-call media-related signaling to the specific SIP trunk. The command midcall-signaling block needs to be configured to enable this behavior. Video escalation and T.38 call flow are rejected when the midcall-signaling block command is configured. This command should be configured only when basic call is the focus and mid-call can be consumed.

Mid Call Codec Preservation

You can enable or disable codec negotiation in the middle of a call. This method defines whether a codec can be negotiated after a call has been initiated.

Behavior of Mid-call Re-INVITE/UPDATE Consumption

The purpose of Mid-call Re-INVITE Consumption is to ensure smooth interoperatibility of supplementary services like audio Hold/Resume and call transfer. This feature should be used as a last resort only when there is no other option in Cisco UBE. This is because configuring this feature can break video-related features. For Delay-offer Re-INVITE, the configured codec will be passed as an offer in 200 message to change the codec, the transcoder is added in the answer.

How to Configure Cisco UBE Mid-call Re-INVITE Consumption

Configuring Passthrough SIP Messages at the Global Level

Perform this task to configure passthrough SIP messages at the global level:

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    voice service voip

    4.    sip

    5.    midcall-signaling passthru media-change

    6.    exit


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example:
    Device> enable
     

    Enables privileged EXEC mode.

    • Enter your password if prompted.
     
    Step 2 configure terminal


    Example:
    Device# configure terminal
     

    Enters global configuration mode.

     
    Step 3 voice service voip


    Example:
    Device(config)# voice service voip
     

    Enters voice service VoIP configuration mode.

     
    Step 4 sip


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

    Enters voice service VoIP SIP configuration mode.

     
    Step 5 midcall-signaling passthru media-change


    Example:
    Device(conf-serv-sip)# midcall-signaling passthru media-change
     

    Passes through SIP messages that involve media change.

     
    Step 6 exit


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

    Exits voice service SIP configuration mode and returns to global configuration mode.

     

    Configuring Passthrough SIP Messages at Dial Peer Level

    Perform this task to configure passthrough SIP messages at the dial-peer level. You need to perform this task at the dial-peer level to consume all media-related mid-call Re-INVITEs/UPDATEs.


    Note


    If the Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature is configured on global and dial-peer level, dial-peer level takes precedence.


    SUMMARY STEPS

      1.    enable

      2.    configure terminal

      3.    dial-peer voice dial-peer tag voip

      4.    voice-class sip mid-call signaling passthru media-change

      5.    exit


    DETAILED STEPS
       Command or ActionPurpose
      Step 1 enable


      Example:
      Device> enable
       

      Enables privileged EXEC mode.

      • Enter your password if prompted.
       
      Step 2 configure terminal


      Example:
      Device# configure terminal
       

      Enters global configuration mode.

       
      Step 3 dial-peer voice dial-peer tag voip


      Example:
      Device(config)# dial-peer voice  2  voip
       

      Enters dial-peer voice configuration mode.

       
      Step 4 voice-class sip mid-call signaling passthru media-change


      Example:
      Device(config-dial-peer)# voice-class sip mid-call signaling passthru media-change
       

      Passes through SIP messages that involve media change.

       
      Step 5 exit


      Example:
      Device(config-dial-peer)# exit
       

      Exits dial-peer voice configuration mode and returns to global configuration mode.

       

      Blocking SIP Messages at the Global Level

      Perform this task to block SIP messages at the global level:

      SUMMARY STEPS

        1.    enable

        2.    configure terminal

        3.    voice service voip

        4.    sip

        5.    midcall-signaling block

        6.    exit


      DETAILED STEPS
         Command or ActionPurpose
        Step 1 enable


        Example:
        Device> enable
         

        Enables privileged EXEC mode.

        • Enter your password if prompted.
         
        Step 2 configure terminal


        Example:
        Device# configure terminal
         

        Enters global configuration mode.

         
        Step 3 voice service voip


        Example:
        Device(config)# voice service voip
         

        Enters voice service VoIP configuration mode.

         
        Step 4 sip


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

        Enters voice service SIP configuration mode.

         
        Step 5 midcall-signaling block


        Example:
        Device(conf-serv-sip)# midcall-signaling block


        Example:

        Blocks all SIP messages during mid-call.

         
         
        Step 6 exit


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

        Exits the current mode.

         

        Blocking SIP Messages at Dial Peer Level

        Perform this task to block SIP messages at dial peer level.


        Note


        If the Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature is configured on global and dial-peer level, dial-peer levels takes precedence.


        SUMMARY STEPS

          1.    enable

          2.    configure terminal

          3.    dial-peer voice dial-peer tag voip

          4.    voice-class sip mid-call signaling block

          5.    exit


        DETAILED STEPS
           Command or ActionPurpose
          Step 1 enable


          Example:
          Device> enable
           

          Enables privileged EXEC mode.

          • Enter your password if prompted.
           
          Step 2 configure terminal


          Example:
          Device# configure terminal
           

          Enters global configuration mode.

           
          Step 3 dial-peer voice dial-peer tag voip


          Example:
          Device(config)# dial-peer voice 2 voip


          Example:

          Enters dial-peer voice configuration mode.

           
           
          Step 4 voice-class sip mid-call signaling block


          Example:
          Device(config-dial-peer)# voice-class sip mid-call signaling block
           

          Blocks all SIP messages during mid-call.

           
          Step 5 exit


          Example:
          Device(config-dial-peer)# exit
           

          Exits the current configuration mode.

           

          Configuring Mid Call Codec Preservation at the Global Level

          SUMMARY STEPS

            1.    enable

            2.    configure terminal

            3.    voice service voip

            4.    sip

            5.    midcall-signaling preserve-codec

            6.    exit


          DETAILED STEPS
             Command or ActionPurpose
            Step 1 enable


            Example:
            Device> enable
             

            Enables privileged EXEC mode.

            • Enter your password if prompted.
             
            Step 2 configure terminal


            Example:
            Device# configure terminal
             

            Enters global configuration mode.

             
            Step 3 voice service voip


            Example:
            Device(config)# voice service voip
             

            Enters voice service VoIP configuration mode.

             
            Step 4 sip


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

            Enters voice service VoIP SIP configuration mode.

             
            Step 5 midcall-signaling preserve-codec


            Example:
            Device(conf-serv-sip)# midcall-signaling preserve-codec
             

            Disables codec negotiation in the middle of a call and preserves the codec negotiated before the call.

             
            Step 6 exit


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

            Exits voice service SIP configuration mode and returns to global configuration mode.

             

            Configuring Mid Call Codec Preservation at the Dial Peer Level

            SUMMARY STEPS

              1.    enable

              2.    configure terminal

              3.    dial-peer voice dial-peer tag voip

              4.    voice-class sip midcall-signaling preserve-codec

              5.    exit


            DETAILED STEPS
               Command or ActionPurpose
              Step 1 enable


              Example:
              Device> enable
               

              Enables privileged EXEC mode.

              • Enter your password if prompted.
               
              Step 2 configure terminal


              Example:
              Device# configure terminal
               

              Enters global configuration mode.

               
              Step 3 dial-peer voice dial-peer tag voip


              Example:
              Device(config)# dial-peer voice  2  voip
               

              Enters dial-peer voice configuration mode.

               
              Step 4 voice-class sip midcall-signaling preserve-codec


              Example:
              Device(conf-dial-peer)# voice-class sip midcall-signaling preserve-codec
               

              Disables codec negotiation in the middle of a call and preserves the codec negotiated before the call.

               
              Step 5 exit


              Example:
              Device(config-dial-peer)# exit
               

              Exits dial-peer voice configuration mode and returns to global configuration mode.

               

              Verifying and Troubleshooting Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

              Perform this task to verify mid-call Re-INVITE/UPDATE routing support on Cisco UBE

              SUMMARY STEPS

                1.    enable

                2.    debug ccsip all

                3.    debug voip ccapi inout


              DETAILED STEPS
                Step 1   enable

                Enables privileged EXEC mode.



                Example:
                Device> enable
                
                Step 2   debug ccsip all

                Use this command to enable SIP related debugging.



                Example:
                Device# debug ccsip all
                Received:
                INVITE sip:5555555555@[2208:1:1:1:1:1:1:1118]:5060 SIP/2.0
                Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1115]:5060;branch=z9hG4bK83AE3
                Remote-Party-ID: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;party=calling;screen=no;privacy=off
                From: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;tag=627460F0-1259
                To: <sip:5555555555@[2208:1:1:1:1:1:1:1118]>
                Date: Tue, 01 Mar 2011 08:49:48 GMT
                Call-ID: B30FCDEB-431711E0-8EDECB51-E9F6B1F1@2208:1:1:1:1:1:1:1115
                Supported: 100rel,timer,resource-priority,replaces
                Require: sdp-anat
                Min-SE:  1800
                Cisco-Guid: 2948477781-1125585376-2396638033-3925258737
                User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16
                Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
                CSeq: 101 INVITE
                Max-Forwards: 70
                Timestamp: 1298969388
                Contact: <sip:2222222222@[2208:1:1:1:1:1:1:1115]:5060>
                Expires: 180
                Allow-Events: telephone-event
                Content-Type: application/sdp
                Content-Disposition: session;handling=required
                Content-Length: 495
                v=0
                o=CiscoSystemsSIP-GW-UserAgent 7880 7375 IN IP6 2208:1:1:1:1:1:1:1115
                s=SIP Call
                c=IN IP6 2208:1:1:1:1:1:1:1115
                t=0 0
                a=group:ANAT 1 2
                m=audio 17836 RTP/AVP 0 101 19
                c=IN IP6 2208:1:1:1:1:1:1:1115
                a=mid:1                                                
                a=rtpmap:0 PCMU/8000
                a=rtpmap:101 telephone-event/8000
                a=fmtp:101 0-16
                a=rtpmap:19 CN/8000
                a=ptime:20
                m=audio 18938 RTP/AVP 0 101 19
                c=IN IP4 9.45.36.111
                a=mid:2                                                
                a=rtpmap:0 PCMU/8000
                a=rtpmap:101 telephone-event/8000
                a=fmtp:101 0-16
                a=rtpmap:19 CN/8000
                a=ptime:20
                “Received: 
                INVITE sip:2222222222@[2208:1:1:1:1:1:1:1117]:5060 SIP/2.0
                Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1116]:5060;branch=z9hG4bK38ACE
                Remote-Party-ID: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;party=calling;screen=no;privacy=off
                From: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;tag=4FE8C9C-1630
                To: <sip:2222222222@[2208:1:1:1:1:1:1:1117]>;tag=1001045C-992
                Date: Thu, 10 Feb 2011 12:15:08 GMT
                Call-ID: 5DEDB77E-ADC11208-808BE770-8FCACF34@2208:1:1:1:1:1:1:1117
                Supported: 100rel,timer,resource-priority,replaces,sdp-anat
                Min-SE:  1800
                Cisco-Guid: 1432849350-0876876256-2424621905-3925258737
                User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16
                Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
                CSeq: 101 INVITE
                Max-Forwards: 70
                Timestamp: 1297340108
                Contact: <sip:5555555555@[2208:1:1:1:1:1:1:1116]:5060>
                Expires: 180
                Allow-Events: telephone-event
                Content-Type: application/sdp
                Content-Length: 424
                v=0
                o=CiscoSystemsSIP-GW-UserAgent 8002 7261 IN IP6 2208:1:1:1:1:1:1:1116
                s=SIP Call
                c=IN IP6 2208:1:1:1:1:1:1:1116
                t=0 0
                m=image 17278 udptl t38
                c=IN IP6 2208:1:1:1:1:1:1:1116
                a=T38FaxVersion:0
                a=T38MaxBitRate:14400
                a=T38FaxFillBitRemoval:0
                a=T38FaxTranscodingMMR:0
                a=T38FaxTranscodingJBIG:0
                a=T38FaxRateManagement:transferredTCF
                a=T38FaxMaxBuffer:200
                a=T38FaxMaxDatagram:320
                a=T38FaxUdpEC:t38UDPRedundancy
                
                Step 3   debug voip ccapi inout


                Example:
                The voice gateway sets up negotiating capability with the terminating VoIP leg.
                *Mar  1 15:36:05.028: //45/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind: (dstVdbPtr=0x637E
                C1E0, dstCallId=0x2C, srcCallId=0x2D,
                     caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0
                           codec_bytes=20, signal_type=2})
                *Mar  1 15:36:05.028: //45/xxxxxxxxxxxx/CCAPI/
                cc_api_caps_ind:
                 (Playout: mode 0,
                 initial 60,min 40, max 300) 
                

                The capabilities are acknowledged for both call legs.



                Example:
                *Mar  1 15:36:05.028: //45/xxxxxxxxxxxx/CCAPI/cc_api_caps_ack: (dstVdbPtr=0x637E
                C1E0, dstCallId=0x2C, srcCallId=0x2D,
                     caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0
                           codec_bytes=20, signal_type=2, seq_num_start=2944})
                *Mar  1 15:36:05.028: //44/xxxxxxxxxxxx/CCAPI/cc_api_caps_ack: (dstVdbPtr=0x62EC
                61A4, dstCallId=0x2D, srcCallId=0x2C,
                     caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0
                           codec_bytes=20, signal_type=2, seq_num_start=2944})
                *Mar  1 15:36:05.032: //44/xxxxxxxxxxxx/CCAPI/cc_api_voice_mode_event: callID=0x2C
                *Mar  1 15:36:05.032: //44/45F2AAE28044/CCAPI/cc_api_voice_mode_event: Call Pointer =634A430C
                *Mar  1 15:36:05.032: //44/xxxxxxxxxxxx/SSAPP:-1:-1/sess_appl: ev(52=CC_EV_VOICE
                _MODE_DONE), cid(44), disp(0)
                *Mar  1 15:36:05.032: //44/45F2AAE28044/SSAPP:10002:21/ssaTraceSct:
                Router#
                Router# cid(44)st(SSA_CS_ACTIVE)ev(SSA_EV_VOICE_MODE_DONE)
                oldst(SSA_CS_CONFERENCING)cfid(21)csize(2)in(1)fDest(1)
                *Mar  1 15:36:05.032: //44/45F2AAE28044/SSAPP:10002:21/ssaTraceSct: -cid2(45)st2
                (SSA_CS_ACTIVE)oldst2(SSA_CS_ALERT_RCVD)
                *Mar  1 15:36:05.032: //44/45F2AAE28044/SSAPP:10002:21/ssaIgnore: cid(44), st(SS
                A_CS_ACTIVE),oldst(5), ev(52)
                Router#
                Router#! digit punched
                Router#
                

                The phone at the terminating gateway enters digit1.



                Example:
                *Mar  1 15:36:11.204: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin: (dstVdbPtr=0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D,
                    digit=1, digit_begin_flags=0x0, rtp_timestamp=0x0
                    rtp_expiration=0x0, dest_mask=0x2)
                *Mar  1 15:36:11.504: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end: (dstVdbPtr=
                0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D,
                    digit=1,duration=300,xruleCallingTag=0,xruleCalledTag=0, dest_mask=0x2), digit_tone_mode=0
                

                The phone at the terminating gateway enters digit 2.



                Example:
                *Mar  1 15:36:11.604: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin: (dstVdbPt
                r=0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D,
                    digit=2, digit_begin_flags=0x0, rtp_timestamp=0x0
                    rtp_expiration=0x0, dest_mask=0x2)
                *Mar  1 15:36:11.904: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end: (dstVdbPtr=
                0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D,
                    digit=2,duration=300,xruleCallingTag=0,xruleCalledTag=0, dest_mask=0x2), digit_tone_mode=0
                Router#
                Router#
                *Mar  1 15:36:14.476: //-1/xxxxxxxxxxxx/CCAPI/cc_handle_periodic_timer: Calling
                the callback, ccTimerctx - 0x628B6330
                *Mar  1 15:36:14.476: //-1/xxxxxxxxxxxx/CCAPI/ccTimerStart: ccTimerctx - 0x628B6330
                Router#
                Router# !call hung up  The user at the terminating gateway hangs up the call.
                Router#
                

                Configuration Examples for Cisco UBE Mid-call Re-INVITE Consumption

                Example Configuring Passthrough SIP Messages at the Global Level

                The following example shows how to passthrough SIP messages at the global level:

                Device(config)# voice service voip
                Device(conf-voi-serv)# no ip address trusted authenticate
                Device(conf-voi-serv)# allow-connections sip to sip
                Device(conf-voi-serv)# sip
                Device(conf-serv-sip)# midcall-signaling passthru media-change
                

                Example Configuring Passthrough SIP Messages at Dial Peer Level

                The following example shows how to passthrough SIP messages at the dial peer Level:

                dial-peer voice 600 voip
                 destination-pattern 2222222222
                 session protocol sipv2
                 session target ipv4:9.45.38.39:9001
                voice-class sip mid-call signaling passthru media-change
                 incoming called-number 1111111111
                voice-class codec  2 offer-all
                dial-peer voice 400 voip
                 destination-pattern 1111111111
                 session protocol sipv2
                 session target ipv4:9.45.38.39:9000
                 incoming called-number 2222222222
                voice-class codec 1 offer-all

                Example: Blocking SIP Messages at the Global Level

                The following example shows how to block SIP messages at the global Level

                Device(config)#voice service voip
                Device(config-voi-serv)#no ip address trusted authenticate
                Device(config-voi-serv)#allow-connections sip to sip
                Device(config-voi-serv)#sip
                Device(config-serv-sip)#midcall-signaling block

                Example Blocking SIP Messages at Dial Peer Level

                dial-peer voice 107 voip
                 destination-pattern 74000
                 session protocol sipv2
                 session target ipv4:9.45.36.9
                 incoming called-number 84000
                 voice-class codec 1 offer-all
                !
                dial-peer voice 110 voip
                 destination-pattern 84000
                 session protocol sipv2
                 session target ipv4:9.45.35.2
                 incoming called-number 74000
                 voice-class codec 1 offer-all
                 voice-class sip mid-call signaling block
                !

                Example: Configuring Mid Call Codec Preservation at the Global Level

                Example: Configuring Mid Call Codec Preservation at the Global Level

                Device(config)# voice service voip
                Device(conf-voi-serv)# no ip address trusted authenticate
                Device(conf-voi-serv)# allow-connections sip to sip
                Device(conf-voi-serv)# sip
                Device(conf-serv-sip)# midcall-signaling preserve-codec
                

                Example: Configuring Mid Call Codec Preservation at the Dial Peer Level

                Example: Configuring Mid Call Codec Preservation at the Dial Peer Level

                dial-peer voice 107 voip
                 destination-pattern 74000
                 session protocol sipv2
                 session target ipv4:9.45.36.9
                 incoming called-number 84000
                 voice-class codec 1 offer-all
                !
                dial-peer voice 110 voip
                 destination-pattern 84000
                 session protocol sipv2
                 session target ipv4:9.45.35.2
                 incoming called-number 74000
                 voice-class codec 1 offer-all
                 voice-class sip midcall-signaling preserve-codec
                !

                Feature Information for Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

                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.

                Table 1 Feature Information for Cisco UBE Mid-call Re-INVITE /UPDATE Consumption

                Feature Name

                Releases

                Feature Information

                Cisco UBE Mid-call Signalling

                15.3(2)S, 15.3T

                The Cisco UBE Mid-call signalling helps to disables codec negotiation in the middle of a call and preserves the codec negotiated before the call.

                The following keyword was introduced or modified: preserve-codec.

                Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

                15.2(1)T

                The Cisco UBE Mid-call Re-INVITE consumption helps to avoid interoperability issues by consuming the mid-call Re-INVITEs from Cisco UBE.

                In Cisco IOS Release 15.2(1)T, this feature was implemented on the Cisco Unified Border Element.

                The following commands were introduced or modified: midcall-signaling.

                Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

                Cisco IOS XE Release 3.6S

                The Cisco UBE Mid-call Re-INVITE consumption helps to avoid interoperability issues by consuming the mid-call Re-INVITEs from Cisco UBE.

                In Cisco IOS XE Release 3.6S, this feature was implemented on the Cisco Unified Border Element (Enterprise).

                The following commands were introduced or modified: midcall-signaling.