- Cisco Unified Border Element SIP Support
- SIP Core SIP Technology Enhancements
- Reporting End-of-Call Statistics in SIP BYE Message
- Configurable Hostname in Locally Generated SIP Headers
- SIP Parameter Modification
- SIP Session Timer Support
- SIP-to-SIP Basic Functionality for Session Border Controller
- SIP-to-SIP Supplementary Services for Session Border Controller
- Session Refresh with Reinvites
- Configuring Cisco UBE Out-of-dialog OPTIONS Ping for Specified SIP Servers or Endpoints
- Configuring an Error Response Code upon an Out-of-Dialog OPTIONS Ping Failure
- SIP INFO Method for DTMF Tone Generation
- SIP Enhanced 180 Provisional Response Handling
- Configuring SIP 181 Call is Being Forwarded Message
- Expires Timer Reset on Receiving or Sending SIP 183 Message
- SIP UPDATE Message per RFC 3311
- Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Configurable Pass-through of SIP INVITE Parameters
- Conditional Header Manipulation of SIP Headers
- Transparent Tunneling of QSIG and Q.931
- SIP Diversion Header Enhancements
- SIP History INFO
- SIP Ability to Send a SIP Registration Message on a Border Element
- Configurable SIP Parameters via DHCP
- Multiple Registrars on SIP Trunks
- Additional References
- Glossary
- Finding Feature Information
- Prerequisites SIP Enhanced 180 Provisional Response Handling
- Information About SIP Enhanced 180 Provisional Response Handling
- How to Disable the SIP Enhanced 180 Provisional Response Handling Feature
- Verifying SIP Enhanced 180 Provisional Response Handling
- Configuration Examples for SIP - Enhanced 180 Provisional Response Handling
- Feature Information for 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
- Prerequisites SIP Enhanced 180 Provisional Response Handling
- Information About SIP Enhanced 180 Provisional Response Handling
- How to Disable the SIP Enhanced 180 Provisional Response Handling Feature
- Verifying SIP Enhanced 180 Provisional Response Handling
- Configuration Examples for SIP - Enhanced 180 Provisional Response Handling
- Feature Information for SIP Enhanced 180 Provisional Response Handling
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 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:
DETAILED STEPS
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. |
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Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.
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