Cisco Unified Border Element (Enterprise) SIP Support Configuration Guide, Cisco IOS XE Release 3S
SIP Enhanced 180 Provisional Response Handling
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SIP Enhanced 180 Provisional Response Handling

SIP Enhanced 180 Provisional Response Handling

The SIP: Enhanced 180 Provisional Response Handling feature enables early media cut-through on Cisco IOS gateways for Session Initiation Protocol (SIP) 180 response messages.

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 SIP Enhanced 180 Provisional Response Handling

Cisco Unified Border Element

  • Cisco IOS Release 12.2(8)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.

Information About SIP Enhanced 180 Provisional Response Handling

The Session Initiation Protocol (SIP) feature allows you to specify whether 180 messages with Session Description Protocol (SDP) are handled in the same way as 183 responses with SDP. The 180 Ringing message is a provisional or informational response used to indicate that the INVITE message has been received by the user agent and that alerting is taking place. The 183 Session Progress response indicates that information about the call state is present in the message body media information. Both 180 and 183 messages may contain SDP, which allows an early media session to be established prior to the call being answered.

Prior to this feature, Cisco gateways handled a 180 Ringing response with SDP in the same manner as a 183 Session Progress response; that is, the SDP was assumed to be an indication that the far end would send early media. Cisco gateways handled a 180 response without SDP by providing local ringback, rather than early media cut-through. This feature provides the capability to ignore the presence or absence of SDP in 180 messages, and as a result, treat all 180 messages in a uniform manner. The SIP: Enhanced 180 Provisional Response Handling feature allows you to specify which call treatment, early media or local ringback, is provided for 180 responses with SDP:

The table below shows the call treatments available with this feature:

Table 1 Call Treatments with SIP Enhanced 180 Provisional Response Handling

Response Message

SIP Enhanced 180 Provisional Response Handling Status

Treatment

180 response with SDP

Enabled (default)

Early media cut-through

180 response with SDP

Disabled

Local ringback

180 response without SDP

Not affected by the SIP--Enhanced 180 Provisional Response Handlingfeature

Local ringback

183 response with SDP

Not affected by the SIP--Enhanced 180 Provisional Response Handling feature

Early media cut-through

How to Disable the SIP Enhanced 180 Provisional Response Handling Feature

Disabling Early Media Cut-Through

The early media cut-through feature is enabled by default. To disable early media cut-through, perform the following task:

SUMMARY STEPS

    1.    enable

    2.    configure terminal

    3.    interface type number

    4.    sip ua

    5.    disable-early-media 180


DETAILED STEPS
     Command or ActionPurpose
    Step 1 enable


    Example:
    Router> enable
     

    Enables privileged EXEC mode.

    • Enter your password if prompted.
     
    Step 2 configure terminal


    Example:
    Router# configure terminal
     

    Enters global configuration mode.

     
    Step 3 interface type number


    Example:
    Router(config)# ethernet 0/0/0
     

    Configures an interface type and enters interface configuration mode.

     
    Step 4 sip ua


    Example:
    Router(config-sip-ua)# sip ua
     

    Enables SIP UA configuration commands in order to configure the user agent.

     
    Step 5 disable-early-media 180


    Example:
    Router(config-sip-ua)# disable-early-media 180
     

    Disables the gateway’s ability to process SDP in a 180 response as a request for early media cut-through.

     

    Verifying SIP Enhanced 180 Provisional Response Handling

    • To verify the SIP Enhanced 180 Provisional Response Handling feature use the show running configuration or show sip-ua statusor show loggingcommand to display the output.
    • If early media is enabled, which is the default setting, the show running-config output does not show any information related to the new feature.
    • To monitor this feature, use the show sip-ua statistics and show sip-ua status EXEC commands.

    Configuration Examples for SIP - Enhanced 180 Provisional Response Handling

    show running-config Command

    The following is sample output from the show running-configcommand after the disable-early-media 180command was used:

    Router# show running-config
    .
    .
    .
    dial-peer voice 223 pots
     application session
     destination-pattern 223
     port 1/0/0
    !
    gateway 
    !
    sip-ua 
     disable-early-media 180

    show sip-ua status Command

    The following is sample output from the show sip-ua statuscommand after the disable-early-media 180command was used.

    Router# 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):ENABLED 10.0.0.0
    SIP User Agent bind status(media):ENABLED 0.0.0.0
    SIP early-media for 180 responses with SDP:DISABLED
    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
    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 

    show logging Command

    The following is partial sample output from the show logging command. The outgoing gateway is receiving a 180 message with SDP and is configured to ignore the SDP.

    Router# show logging
    Log Buffer (600000 bytes):
    00:12:19:%SYS-5-CONFIG_I:Configured from console by console
    00:12:19:%SYS-5-CONFIG_I:Configured from console by console
    00:12:20:0x639F6EEC :State change from (STATE_NONE, SUBSTATE_NONE)  to 
    (STATE_IDLE, SUBSTATE_NONE)
    00:12:20:****Adding to UAC table
    00:12:20:adding call id 2 to table
    00:12:20: Queued event from SIP SPI :SIPSPI_EV_CC_CALL_SETUP
    00:12:20:CCSIP-SPI-CONTROL: act_idle_call_setup
    00:12:20: act_idle_call_setup:Not using Voice Class Codec
    00:12:20:act_idle_call_setup:preferred_codec set[0] type :g711ulaw 
    bytes:160
    00:12:20:sipSPICopyPeerDataToCCB:From CLI:Modem NSE payload = 100, 
    Passthrough = 0,Modem relay = 0, Gw-Xid = 1
    SPRT latency 200, SPRT Retries = 12, Dict Size = 1024
    String Len = 32, Compress dir = 3
    00:12:20:sipSPICanSetFallbackFlag - Local Fallback is not active
    00:12:20:****Deleting from UAC table
    00:12:20:****Adding to UAC table
    00:12:20: Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION
    00:12:20:0x639F6EEC :State change from (STATE_IDLE, SUBSTATE_NONE)  to 
    (STATE_IDLE, SUBSTATE_CONNECTING)
    00:12:20:0x639F6EEC :State change from (STATE_IDLE, 
    SUBSTATE_CONNECTING)  to (STATE_IDLE, SUBSTATE_CONNECTING)
    00:12:20:sipSPIUsetBillingProfile:sipCallId for billing records = 
    41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42
    00:12:20:CCSIP-SPI-CONTROL: act_idle_connection_created
    00:12:20:CCSIP-SPI-CONTROL: act_idle_connection_created:Connid(1) 
    created to 172.31.1.15:5060, local_port 57838
    00:12:20:CCSIP-SPI-CONTROL: sipSPIOutgoingCallSDP
    00:12:20:sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.1.1.42
    00:12:20:sipSPIReserveRtpPort:reserved port 18978 for stream 1
    00:12:20: convert_codec_bytes_to_ptime:Values :Codec:g711ulaw 
    codecbytes :160, ptime:20
    00:12:20:sip_generate_sdp_xcaps_list:Modem Relay disabled. X-cap not 
    needed
    00:12:20:Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off
    00:12:20:sipSPIAddLocalContact
    00:12:20: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
    00:12:20:sip_stats_method
    00:12:20:sipSPIProcessRtpSessions
    00:12:20:sipSPIAddStream:Adding stream 1 (callid 2) to the VOIP RTP 
    library
    00:12:20:sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.1.1.42
    00:12:20:sipSPIUpdateRtcpSession:for m-line 1
    00:12:20:sipSPIUpdateRtcpSession:rtcp_session info
    laddr = 10.1.1.42, lport = 18978, raddr = 0.0.0.0, 
    rport=0, do_rtcp=FALSE
    src_callid = 2, dest_callid = -1
    00:12:20:sipSPIUpdateRtcpSession:No rtp session, creating a new one
    00:12:20:sipSPIAddStream:In State Idle
    00:12:20:act_idle_connection_created:Transaction active. Facilities will 
    be queued.
    00:12:20:0x639F6EEC :State change from (STATE_IDLE, 
    SUBSTATE_CONNECTING)  to (STATE_SENT_INVITE, SUBSTATE_NONE)
    00:12:20:Sent:
    INVITE sip:222@172.31.1.15:5060 SIP/2.0
    Via:SIP/2.0/UDP  10.1.1.42:5060
    From:"111" <sip:111@172.31.1.42>;tag=B4DC4-9E1
    To:<sip:222@172.31.1.15>
    Date:Mon, 01 Mar 1993 00:12:20 GMT
    Call-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42
    Supported:timer
    Min-SE: 1800
    Cisco-Guid:1096070726-351277516-2147659648-3567923539
    User-Agent:Cisco-SIPGateway/IOS-12.x
    Allow:INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
    NOTIFY, INFO
    CSeq:101 INVITE
    Max-Forwards:6
    Remote-Party-ID:<sip:111@172.31.1.42>;party=calling;screen=no;privacy=off
    Timestamp:730944740
    Contact:<sip:111@172.31.1.42:5060>
    Expires:180
    Allow-Events:telephone-event
    Content-Type:application/sdp
    Content-Length:230
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4629 354 IN IP4 172.31.1.42
    s=SIP Call
    c=IN IP4 172.31.1.42
    t=0 0
    m=audio 18978 RTP/AVP 0 100
    c=IN IP4 10.1.1.42
    a=rtpmap:0 PCMU/8000
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 192-194
    a=ptime:20
    00:12:21:Received:
    SIP/2.0 100 Trying
    Via:SIP/2.0/UDP  10.1.1.42:5060
    From:"111" <sip:111@172.31.1.42>;tag=B4DC4-9E1
    To:<sip:222@172.31.1.15>;tag=442AC-22
    Date:Wed, 16 Feb 2000 18:19:56 GMT
    Call-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42
    Timestamp:730944740
    Server:Cisco-SIPGateway/IOS-12.x
    CSeq:101 INVITE
    Allow-Events:telephone-event
    Content-Length:0
    00:12:21:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:
    10.1.1.15:5060
    00:12:21:CCSIP-SPI-CONTROL: act_sentinvite_new_message
    00:12:21:CCSIP-SPI-CONTROL: sipSPICheckResponse
    00:12:21:sip_stats_status_code
    00:12:21: Roundtrip delay 420 milliseconds for method INVITE
    00:12:21:0x639F6EEC :State change from (STATE_SENT_INVITE, 
    SUBSTATE_NONE)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
    00:12:21:Received:
    SIP/2.0 180 Ringing
    Via:SIP/2.0/UDP  10.1.1.42:5060
    From:"111" <sip:111@10.1.1.42>;tag=B4DC4-9E1
    To:<sip:222@172.31.1.15>;tag=442AC-22
    Date:Wed, 16 Feb 2000 18:19:56 GMT
    Call-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42
    Timestamp:730944740
    Server:Cisco-SIPGateway/IOS-12.x
    CSeq:101 INVITE
    Allow-Events:telephone-event
    Contact:<sip:222@172.31.1.59:5060>
    Record-Route:<sip:222@10.1.1.15:5060;maddr=10.1.1.15>
    Content-Length:230
    Content-Type:application/sdp
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4629 354 IN IP4 10.1.1.42
    s=SIP Call
    c=IN IP4 10.1.1.42
    t=0 0
    m=audio 18978 RTP/AVP 0 100
    c=IN IP4 10.1.1.42
    a=rtpmap:0 PCMU/8000
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 192-194
    a=ptime:20
    00:12:21:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:
    10.1.1.15:5060
    00:12:21:CCSIP-SPI-CONTROL: act_recdproc_new_message
    00:12:21:CCSIP-SPI-CONTROL: act_recdproc_new_message_response
    00:12:21:CCSIP-SPI-CONTROL: sipSPICheckResponse
    00:12:21:sip_stats_status_code
    00:12:21: Roundtrip delay 496 milliseconds for method INVITE
    00:12:21:CCSIP-SPI-CONTROL: act_recdproc_new_message_response :Early 
    media disabled for 180:Ignoring SDP if present
    00:12:21:HandleSIP1xxRinging:SDP in 180 will be ignored if present: No 
    early media cut through
    00:12:21:HandleSIP1xxRinging:SDP Body either absent or ignored in 180 
    RINGING:- would wait for 200 OK to do negotiation.
    00:12:21:HandleSIP1xxRinging:MediaNegotiation expected in 200 OK
    00:12:21:sipSPIGetGtdBody:No valid GTD body found.
    00:12:21:sipSPICreateRawMsg:No GTD passed.
    00:12:21:0x639F6EEC :State change from (STATE_RECD_PROCEEDING, 
    SUBSTATE_PROCEEDING_PROCEEDING)  to (STATE_RECD_PROCEEDING, 
    SUBSTATE_PROCEEDING_ALERTING)
    00:12:21:HandleSIP1xxRinging:Transaction Complete. Lock on Facilities 
    released.
    00:12:22:Received:
    SIP/2.0 200 OK
    Via:SIP/2.0/UDP  10.1.1.42:5060
    From:"111" <sip:111@10.1.1.42>;tag=B4DC4-9E1
    To:<sip:222@10.1.1.15>;tag=442AC-22
    Date:Wed, 16 Feb 2000 18:19:56 GMT
    Call-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42
    Timestamp:730944740
    Server:Cisco-SIPGateway/IOS-12.x
    CSeq:101 INVITE
    Allow:INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
    NOTIFY, INFO
    Allow-Events:telephone-event
    Contact:<sip:222@10.1.1.59:5060>
    Record-Route:<sip:222@10.1.1.15:5060;maddr=10.1.1.15>
    Content-Type:application/sdp
    Content-Length:231
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9600 4816 IN IP4 10.1.1.59
    s=SIP Call
    c=IN IP4 10.1.1.59
    t=0 0
    m=audio 19174 RTP/AVP 0 100
    c=IN IP4 10.1.1.59
    a=rtpmap:0 PCMU/8000
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 192-194
    a=ptime:20
    

    Feature Information for SIP Enhanced 180 Provisional Response Handling

    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 Information Table for the ISR

    Table 2 Feature Information for SIP :Enhanced 180 Provisional Response Handling

    Feature Name

    Releases

    Feature Information

    SIP - Enhanced 180 Provisional Response Handling

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

    The Session Initiation Protocol (SIP) Enhanced 180 Provisional Response Handling feature provides the ability to enable or disable early media cut-through on Cisco IOS gateways for SIP 180 response messages.

    The following commands were introduced or modified: disable-early-media 180 and show sip-ua status.

    Feature Information Table for the ASR

    Table 3 Feature Information for SIP: Enhanced 180 Provisional Response Handling

    Feature Name

    Releases

    Feature Information

    SIP - Enhanced 180 Provisional Response Handling

    Cisco IOS XE Release 2.5

    The Session Initiation Protocol (SIP) Enhanced 180 Provisional Response Handling feature provides the ability to enable or disable early media cut-through on Cisco IOS gateways for SIP 180 response messages.

    The following commands were introduced or modified: disable-early-media 180 and show sip-ua status.