Table Of Contents
SIP—Enhanced 180 Provisional Response Handling
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
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.
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:
Table 1 shows the call treatments available with this feature:
How to Disable the SIP Enhanced 180 Provisional Response Handling Feature
This section describes the configuration tasks for the SIP Enhanced 180 Provisional Response Handling feature:
•Disabling Early Media Cut-Through (optional)
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 STEPSVerifying 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 status or show logging command 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
This section displays sample outputs from the following show commands:
show running-config Command
The following is sample output from the show running-config command after the disable-early-media 180 command was used:
Router# show running-config
...dial-peer voice 223 potsapplication sessiondestination-pattern 223port 1/0/0!gateway!sip-uadisable-early-media 180show sip-ua status Command
The following is sample output from the show sip-ua status command after the disable-early-media 180 command was used.
Router# show sip-ua status
SIP User Agent StatusSIP User Agent for UDP :ENABLEDSIP User Agent for TCP :ENABLEDSIP User Agent bind status(signaling):ENABLED 10.0.0.0SIP User Agent bind status(media):ENABLED 0.0.0.0SIP early-media for 180 responses with SDP:DISABLEDSIP max-forwards :6SIP DNS SRV version:2 (rfc 2782)NAT Settings for the SIP-UARole in SDP:NONECheck media source packets:DISABLEDRedirection (3xx) message handling:ENABLEDSDP application configuration:Version line (v=) requiredOwner line (o=) requiredTimespec line (t=) requiredMedia supported:audio imageNetwork types supported:INAddress types supported:IP4Transport types supported:RTP/AVP udptlshow 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 console00:12:19:%SYS-5-CONFIG_I:Configured from console by console00:12:20:0x639F6EEC :State change from (STATE_NONE, SUBSTATE_NONE) to(STATE_IDLE, SUBSTATE_NONE)00:12:20:****Adding to UAC table00:12:20:adding call id 2 to table00:12:20: Queued event from SIP SPI :SIPSPI_EV_CC_CALL_SETUP00:12:20:CCSIP-SPI-CONTROL: act_idle_call_setup00:12:20: act_idle_call_setup:Not using Voice Class Codec00:12:20:act_idle_call_setup:preferred_codec set[0] type :g711ulawbytes:16000:12:20:sipSPICopyPeerDataToCCB:From CLI:Modem NSE payload = 100,Passthrough = 0,Modem relay = 0, Gw-Xid = 1SPRT latency 200, SPRT Retries = 12, Dict Size = 1024String Len = 32, Compress dir = 300:12:20:sipSPICanSetFallbackFlag - Local Fallback is not active00:12:20:****Deleting from UAC table00:12:20:****Adding to UAC table00:12:20: Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION00:12:20:0x639F6EEC :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@172.31.1.4200:12:20:CCSIP-SPI-CONTROL: act_idle_connection_created00:12:20:CCSIP-SPI-CONTROL: act_idle_connection_created:Connid(1)created to 172.31.1.15:5060, local_port 5783800:12:20:CCSIP-SPI-CONTROL: sipSPIOutgoingCallSDP00:12:20:sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.1.1.4200:12:20:sipSPIReserveRtpPort:reserved port 18978 for stream 100:12:20: convert_codec_bytes_to_ptime:Values :Codec:g711ulawcodecbytes :160, ptime:2000:12:20:sip_generate_sdp_xcaps_list:Modem Relay disabled. X-cap notneeded00:12:20:Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off00:12:20:sipSPIAddLocalContact00:12:20: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE00:12:20:sip_stats_method00:12:20:sipSPIProcessRtpSessions00:12:20:sipSPIAddStream:Adding stream 1 (callid 2) to the VOIP RTPlibrary00:12:20:sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.1.1.4200:12:20:sipSPIUpdateRtcpSession:for m-line 100:12:20:sipSPIUpdateRtcpSession:rtcp_session infoladdr = 10.1.1.42, lport = 18978, raddr = 0.0.0.0,rport=0, do_rtcp=FALSEsrc_callid = 2, dest_callid = -100:12:20:sipSPIUpdateRtcpSession:No rtp session, creating a new one00:12:20:sipSPIAddStream:In State Idle00:12:20:act_idle_connection_created:Transaction active. Facilities willbe 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/2.0Via:SIP/2.0/UDP 10.1.1.42:5060From:"111" <sip:111@172.31.1.42>;tag=B4DC4-9E1To:<sip:222@172.31.1.15>Date:Mon, 01 Mar 1993 00:12:20 GMTCall-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42Supported:timerMin-SE: 1800Cisco-Guid:1096070726-351277516-2147659648-3567923539User-Agent:Cisco-SIPGateway/IOS-12.xAllow:INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,NOTIFY, INFOCSeq:101 INVITEMax-Forwards:6Remote-Party-ID:<sip:111@172.31.1.42>;party=calling;screen=no;privacy=offTimestamp:730944740Contact:<sip:111@172.31.1.42:5060>Expires:180Allow-Events:telephone-eventContent-Type:application/sdpContent-Length:230v=0o=CiscoSystemsSIP-GW-UserAgent 4629 354 IN IP4 172.31.1.42s=SIP Callc=IN IP4 172.31.1.42t=0 0m=audio 18978 RTP/AVP 0 100c=IN IP4 10.1.1.42a=rtpmap:0 PCMU/8000a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194a=ptime:2000:12:21:Received:SIP/2.0 100 TryingVia:SIP/2.0/UDP 10.1.1.42:5060From:"111" <sip:111@172.31.1.42>;tag=B4DC4-9E1To:<sip:222@172.31.1.15>;tag=442AC-22Date:Wed, 16 Feb 2000 18:19:56 GMTCall-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42Timestamp:730944740Server:Cisco-SIPGateway/IOS-12.xCSeq:101 INVITEAllow-Events:telephone-eventContent-Length:000:12:21:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:10.1.1.15:506000:12:21:CCSIP-SPI-CONTROL: act_sentinvite_new_message00:12:21:CCSIP-SPI-CONTROL: sipSPICheckResponse00:12:21:sip_stats_status_code00:12:21: Roundtrip delay 420 milliseconds for method INVITE00:12:21:0x639F6EEC :State change from (STATE_SENT_INVITE,SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)00:12:21:Received:SIP/2.0 180 RingingVia:SIP/2.0/UDP 10.1.1.42:5060From:"111" <sip:111@10.1.1.42>;tag=B4DC4-9E1To:<sip:222@172.31.1.15>;tag=442AC-22Date:Wed, 16 Feb 2000 18:19:56 GMTCall-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42Timestamp:730944740Server:Cisco-SIPGateway/IOS-12.xCSeq:101 INVITEAllow-Events:telephone-eventContact:<sip:222@172.31.1.59:5060>Record-Route:<sip:222@10.1.1.15:5060;maddr=10.1.1.15>Content-Length:230Content-Type:application/sdpv=0o=CiscoSystemsSIP-GW-UserAgent 4629 354 IN IP4 10.1.1.42s=SIP Callc=IN IP4 10.1.1.42t=0 0m=audio 18978 RTP/AVP 0 100c=IN IP4 10.1.1.42a=rtpmap:0 PCMU/8000a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194a=ptime:2000:12:21:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:10.1.1.15:506000:12:21:CCSIP-SPI-CONTROL: act_recdproc_new_message00:12:21:CCSIP-SPI-CONTROL: act_recdproc_new_message_response00:12:21:CCSIP-SPI-CONTROL: sipSPICheckResponse00:12:21:sip_stats_status_code00:12:21: Roundtrip delay 496 milliseconds for method INVITE00:12:21:CCSIP-SPI-CONTROL: act_recdproc_new_message_response :Earlymedia disabled for 180:Ignoring SDP if present00:12:21:HandleSIP1xxRinging:SDP in 180 will be ignored if present: Noearly media cut through00:12:21:HandleSIP1xxRinging:SDP Body either absent or ignored in 180RINGING:- would wait for 200 OK to do negotiation.00:12:21:HandleSIP1xxRinging:MediaNegotiation expected in 200 OK00:12:21: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 Facilitiesreleased.00:12:22:Received:SIP/2.0 200 OKVia:SIP/2.0/UDP 10.1.1.42:5060From:"111" <sip:111@10.1.1.42>;tag=B4DC4-9E1To:<sip:222@10.1.1.15>;tag=442AC-22Date:Wed, 16 Feb 2000 18:19:56 GMTCall-ID:41585FCE-14F011CC-8005AF80-D4AA3153@172.31.1.42Timestamp:730944740Server:Cisco-SIPGateway/IOS-12.xCSeq:101 INVITEAllow:INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,NOTIFY, INFOAllow-Events:telephone-eventContact:<sip:222@10.1.1.59:5060>Record-Route:<sip:222@10.1.1.15:5060;maddr=10.1.1.15>Content-Type:application/sdpContent-Length:231v=0o=CiscoSystemsSIP-GW-UserAgent 9600 4816 IN IP4 10.1.1.59s=SIP Callc=IN IP4 10.1.1.59t=0 0m=audio 19174 RTP/AVP 0 100c=IN IP4 10.1.1.59a=rtpmap:0 PCMU/8000a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194a=ptime:20