Cisco Unified Border Element (Enterprise) Protocol-Independent Features and Setup Configuration Guide, Cisco IOS XE Release 3S
DTMF Events through SIP Signaling
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DTMF Events through SIP Signaling

DTMF Events through SIP Signaling

The DTMF Events through SIP Signaling feature provides the following:

  • DTMF event notification for SIP messages.
  • Capability of receiving hookflash event notification through the SIP NOTIFY method.
  • Third-party call control, or other signaling mechanisms, to provide enhanced services, such as calling card and messaging services.
  • Communication with the application outside of the media connection.

The DTMF Events through SIP Signaling feature allows telephone event notifications to be sent through SIP NOTIFY messages, using the SIP SUBSCRIBE/NOTIFY method as defined in the Internet Engineering Task Force (IETF) draft, SIP-Specific Event Notification.

The feature also supports sending DTMF notifications based on the IETF draft: Signaled Telephony Events in the Session Initiation Protocol (SIP) (draft-mahy-sip-signaled-digits-01.txt).

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 DTMF Events through SIP Signaling

Cisco Unified Border Element

  • Cisco IOS Release 12.2(11)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 2.5 or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Restrictions for DTMF Events through SIP Signaling

The DTMF Events through SIP Signaling feature adds support for sending telephone-event notifications via SIP NOTIFY messages from a SIP gateway. The events for which notifications are sent out are DTMF events from the local Plain Old Telephone Service (POTS) interface on the gateway. Notifications are not sent for DTMF events received in the Real-Time Transport Protocol (RTP) stream from the recipient user agent.

Configuring DTMF Events through SIP Signaling

To configure the DTMF Events through SIP Signaling feature, perform the following steps.

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    sip-ua

    4.    timers notify number

    5.    retry notify number

    6.    exit


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example:
    Device> enable
     

    Enters privileged EXEC mode or any other security level set by a system administrator.

    • Enter your password if prompted.
     
    Step 2 configure terminal


    Example:
    Device# configure terminal
     

    Enters global configuration mode.

     
    Step 3 sip-ua


    Example:
    Device(config)# sip-ua
     

    Enters SIP user-agent configuration mode.

     
    Step 4 timers notify number


    Example:
    Device(config-sip-ua)# timers notify 100
     

    Sets the amount of time that the user agent waits before retransmitting the Notify message. The argument is as follows:

    • number --Time, in milliseconds, to wait before retransmitting. Range: 100 to 1000. Default: 500.
     
    Step 5 retry notify number


    Example:
    Device(config-sip-ua)# retry notify 6
     

    Sets the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request. The argument is as follows:

    • number --Number of retries. Range: 1 to 10. Default: 10.
     
    Step 6 exit


    Example:
    Device(config-sip-ua)# exit
     

    Exits the current mode.

     

    Verifying SIP DTMF Support

    To verify SIP DTMF support, perform the following steps as appropriate (commands are listed in alphabetical order).

    SUMMARY STEPS

      1.    show running-config

      2.    show sip-ua retry

      3.    show sip-ua statistics

      4.    show sip-ua status

      5.    show sip-ua timers

      6.    show voip rtp connections

      7.    show sip-ua calls


    DETAILED STEPS
      Step 1   show running-config

      Use this command to show dial-peer configurations.

      The following sample output shows that the dtmf-relay sip-notify command is configured in dial peer 123:



      Example:
      Device# show running-config
      .
      .
      .
      dial-peer voice 123 voip
       destination-pattern [12]...
       monitor probe icmp-ping
       session protocol sipv2
       session target ipv4:10.8.17.42
       dtmf-relay sip-notify
      

      The following sample output shows that DTMF relay and NTE are configured on the dial peer.



      Example:
      Device# show running-config
      !
      dial-peer voice 1000 pots
       destination-pattern 4961234
       port 1/0/0
      !
      dial-peer voice 2000 voip
       application session
       destination-pattern 4965678
       session protocol sipv2
       session target ipv4:192.0.2.34
       dtmf-relay rtp-nte
      ! RTP payload type value = 101 (default)
      !
      dial-peer voice 3000 voip
       application session
       destination-pattern 2021010101
       session protocol sipv2
       session target ipv4:192.0.2.34
       dtmf-relay rtp-nte
       rtp payload-type nte 110
      ! RTP payload type value = 110 (user assigned)
      !
      
      Step 2   show sip-ua retry

      Use this command to display SIP retry statistics.



      Example:
      Device# show sip-ua retry
      SIP UA Retry Values
      invite retry count = 6 response retry count = 1
      bye retry count = 1 cancel retry count = 1
      prack retry count = 10 comet retry count = 10
      reliable 1xx count = 6 notify retry count = 10
      
      Step 3   show sip-ua statistics

      Use this command to display response, traffic, and retry SIP statistics.

      Tip   

      To reset counters for the show sip-ua statisticsdisplay, use the clear sip-ua statistics command.



      Example:
      Device# show sip-ua statistics
      SIP Response Statistics (Inbound/Outbound)
      Informational:
      Trying 4/2, Ringing 2/1,
      Forwarded 0/0, Queued 0/0,
      SessionProgress 0/0
      Success:
      OkInvite 1/2, OkBye 0/1,
      OkCancel 1/0, OkOptions 0/0,
      OkPrack 2/0, OkPreconditionMet 0/0,
      OkNotify 1/0, 202Accepted 0/1
      Redirection (Inbound only):
      MultipleChoice 0, MovedPermanently 0,
      MovedTemporarily 0, SeeOther 0,
      UseProxy 0, AlternateService 0
      Client Error:
      BadRequest 0/0, Unauthorized 0/0,
      PaymentRequired 0/0, Forbidden 0/0,
      NotFound 0/0, MethodNotAllowed 0/0,
      NotAcceptable 0/0, ProxyAuthReqd 0/0,
      ReqTimeout 0/0, Conflict 0/0, Gone 0/0,
      LengthRequired 0/0, ReqEntityTooLarge 0/0,
      ReqURITooLarge 0/0, UnsupportedMediaType 0/0,
      BadExtension 0/0, TempNotAvailable 0/0,
      CallLegNonExistent 0/0, LoopDetected 0/0,
      TooManyHops 0/0, AddrIncomplete 0/0,
      Ambiguous 0/0, BusyHere 0/0
      RequestCancel 1/0, NotAcceptableMedia 0/0
      Server Error:
      InternalError 0/1, NotImplemented 0/0,
      BadGateway 0/0, ServiceUnavail 0/0,
      GatewayTimeout 0/0, BadSipVer 0/0,
      PreCondFailure 0/0
      Global Failure:
      BusyEverywhere 0/0, Decline 0/0,
      NotExistAnywhere 0/0, NotAcceptable 0/0
      SIP Total Traffic Statistics (Inbound/Outbound) /* Traffic Statistics
      Invite 3/2, Ack 3/2, Bye 1/0,
      Cancel 0/1, Options 0/0,
      Prack 0/2, Comet 0/0,
      Notify 0/1, Refer 1/0
      Retry Statistics 							/* Retry Statistics
      Invite 0, Bye 0, Cancel 0, Response 0,
      Prack 0, Comet 0, Reliable1xx 0, Notify 0
      

      Following is sample output verifying configuration of the SIP INFO Method for DTMF Tone Generation feature:



      Example:
      Device# show sip-ua statistics
      SIP Response Statistics (Inbound/Outbound)
      Informational:
      Trying 1/1, Ringing 0/0,
      Forwarded 0/0, Queued 0/0,
      SessionProgress 0/1
      Success:
      OkInvite 0/1, OkBye 1/0,
      OkCancel 0/0, OkOptions 0/0,
      OkPrack 0/0, OkPreconditionMet 0/0
      OkSubscibe 0/0, OkNotify 0/0,
      OkInfo 0/0, 202Accepted 0/0
      Redirection (Inbound only):
      MultipleChoice 0, MovedPermanently 0,
      MovedTemporarily 0, SeeOther 0,
      UseProxy 0, AlternateService 0
      Client Error:
      BadRequest 0/0, Unauthorized 0/0,
      PaymentRequired 0/0, Forbidden 0/0,
      NotFound 0/0, MethodNotAllowed 0/0,
      NotAcceptable 0/0, ProxyAuthReqd 0/0,
      ReqTimeout 0/0, Conflict 0/0, Gone 0/0,
      LengthRequired 0/0, ReqEntityTooLarge 0/0,
      ReqURITooLarge 0/0, UnsupportedMediaType 0/0,
      BadExtension 0/0, TempNotAvailable 0/0,
      CallLegNonExistent 0/0, LoopDetected 0/0,
      TooManyHops 0/0, AddrIncomplete 0/0,
      Ambiguous 0/0, BusyHere 0/0,
      BadEvent 0/0
      Server Error:
      InternalError 0/0, NotImplemented 0/0,
      BadGateway 0/0, ServiceUnavail 0/0,
      GatewayTimeout 0/0, BadSipVer 0/0
      Global Failure:
      BusyEverywhere 0/0, Decline 0/0,
      NotExistAnywhere 0/0, NotAcceptable 0/0
      SIP Total Traffic Statistics (Inbound/Outbound)
          Invite 0/0, Ack 0/0, Bye 0/0,
          Cancel 0/0, Options 0/0,
          Prack 0/0, Comet 0/0,
          Subscribe 0/0, Notify 0/0,
          Refer 0/0, Info 0/0
      Retry Statistics
      Invite 0, Bye 0, Cancel 0, Response 0, Notify 0
      
      Step 4   show sip-ua status

      Use this command to display status for the SIP user agent.



      Example:
      Device# show sip-ua status
      SIP User Agent Status
      SIP User Agent for UDP : ENABLED
      SIP User Agent for TCP : ENABLED
      SIP User Agent bind status(signaling): DISABLED
      SIP User Agent bind status(media): DISABLED
      SIP max-forwards : 6
      SIP DNS SRV version: 2 (rfc 2782)
      SDP application configuration:
       Version line (v=) required
       Owner line (o=) required
       Session name line (s=) required
       Timespec line (t=) required
       Media supported: audio image
       Network types supported: IN
       Address types supported: IP4
       Transport types supported: RTP/AVP udptl
      

      The following sample output shows that the time interval between consecutive NOTIFY messages for a telephone event is the default of 2000 ms:



      Example:
      Device# show sip-ua status
      SIP User Agent Status
      SIP User Agent for UDP : ENABLED
      SIP User Agent for TCP : ENABLED
      SIP User Agent bind status(signaling): DISABLED
      SIP User Agent bind status(media): DISABLED
      SIP early-media for 180 responses with SDP: ENABLED
      SIP max-forwards : 6
      SIP DNS SRV version: 2 (rfc 2782)
      NAT Settings for the SIP-UA
      Role in SDP: NONE
      Check media source packets: DISABLED
      Maximum duration for a telephone-event in NOTIFYs: 2000 ms
      SIP support for ISDN SUSPEND/RESUME: ENABLED
      Redirection (3xx) message handling: ENABLED
       SDP application configuration:
       Version line (v=) required
       Owner line (o=) required
       Timespec line (t=) required
       Media supported: audio image
       Network types supported: IN
       Address types supported: IP4
       Transport types supported: RTP/AVP udptl
      

      The following sample output shows configuration of the SIP INFO Method for DTMF Tone Generation feature:



      Example:
      Device# show sip-ua status
      SIP User Agent Status
      SIP User Agent for UDP : ENABLED
      SIP User Agent for TCP : ENABLED
      SIP User Agent bind status(signaling): DISABLED
      SIP User Agent bind status(media): DISABLED
      SIP max-forwards : 6
      SIP DNS SRV version: 2 (rfc 2782)
      SDP application configuration:
       Version line (v=) required
       Owner line (o=) required
       Session name line (s=) required
       Timespec line (t=) required
       Media supported: audio image
       Network types supported: IN
       Address types supported: IP4
       Transport types supported: RTP/AVP udptl
      
      Step 5   show sip-ua timers

      Use this command to display the current settings for SIP user-agent timers.



      Example:
      Device# show sip-ua timers
      SIP UA Timer Values (millisecs)
      trying 500, expires 300000, connect 500, disconnect 500
      comet 500, prack 500, rel1xx 500, notify 500
      Step 6   show voip rtp connections

      Use this command to show local and remote Calling ID and IP address and port information.

      Step 7   show sip-ua calls

      Use this command to ensure the DTMF method is SIP-KPML.

      The following sample output shows that the DTMF method isSIP-KPML.



      Example:
      Device# show sip-ua calls
      SIP UAC CALL INFO
      Call 1
      SIP Call ID                : 57633F68-2BE011D6-8013D46B-B4F9B5F6@172.18.193.251
         State of the call       : STATE_ACTIVE (7)
         Substate of the call    : SUBSTATE_NONE (0)
         Calling Number          : 
         Called Number           : 8888
         Bit Flags               : 0xD44018 0x100 0x0
         CC Call ID              : 6
         Source IP Address (Sig ): 192.0.2.1
         Destn SIP Req Addr:Port : 192.0.2.2:5060
         Destn SIP Resp Addr:Port: 192.0.2.3:5060
         Destination Name        : 192.0.2.4.250
         Number of Media Streams : 1
         Number of Active Streams: 1
         RTP Fork Object         : 0x0
         Media Mode              : flow-through
         Media Stream 1
           State of the stream      : STREAM_ACTIVE
           Stream Call ID           : 6
           Stream Type              : voice-only (0)
           Negotiated Codec         : g711ulaw (160 bytes)
      	Codec Payload Type       : 0 
           Negotiated Dtmf-relay    : sip-kpml
           Dtmf-relay Payload Type  : 0
           Media Source IP Addr:Port: 192.0.2.5:17576
           Media Dest IP Addr:Port  : 192.0.2.6:17468
           Orig Media Dest IP Addr:Port : 0.0.0.0:0
         Number of SIP User Agent Client(UAC) calls: 1
      SIP UAS CALL INFO
         Number of SIP User Agent Server(UAS) calls: 0

      Troubleshooting Tips

      • To enable debugging for RTP named-event packets, use the debug voip rtp command.
      • To enable KPML debugs, use the debug kpml command.
      • To enable SIP debugs, use the debug ccsip command.
      • Collect debugs while the call is being established and during digit presses.
      • If an established call is not sending digits through KPML, use the show sip-ua calls command to ensure SIP-KPML is included in the negotiation process.

      Feature Information for DTMF Events through SIP Signaling

      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 Configuring DTMF Events through SIP Signaling

      Feature Name

      Releases

      Feature Information

      DTMF Events through SIP Signaling

      12.2(11)T 12.2(8)YN 12.2(15)T 12.2(11)YV 12.2(11)T,

      The DTMF Events through SIP Signaling feature provides the following:

      • DTMF event notification for SIP messages.
      • Capability of receiving hookflash event notification through the SIP NOTIFY method.
      • Third-party call control, or other signaling mechanisms, to provide enhanced services, such as calling card and messaging services.
      • Communication with the application outside of the media connection.

      The following commands were introduced or modified: timers notify and retry notify.

      DTMF Events through SIP Signaling

      Cisco IOS XE Release 2.5

      The DTMF Events through SIP Signaling feature provides the following:

      • DTMF event notification for SIP messages.
      • Capability of receiving hookflash event notification through the SIP NOTIFY method.
      • Third-party call control, or other signaling mechanisms, to provide enhanced services, such as calling card and messaging services.
      • Communication with the application outside of the media connection.

      The following commands were introduced or modified: timers notify and retry notify.