SIP Reliable Provisional Response on CUBE and CUCM Configuration Example

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SIP Reliable Provisional Response on CUBE and CUCM Configuration Example Document ID: 116086 Contributed by Robin Cai, Cisco TAC Engineer. May 16, 2013 Contents Introduction Prerequisites Requirements Components Used Background Information CUBE Configuration CUCM Configuration Typical SIP Messages Troubleshooting Related Information Introduction This document describes how the Session Initiation Protocol (SIP) reliable provisional response feature works and how to confiugred it on Cisco Unified Border Element (CUBE) and Cisco Unified Communications Manager (CUCM). Prerequisites Requirements Cisco recommends that you have knowledge of these topics: Cisco Unified Border Element (CUBE) Enterprise Cisco Unified Communications Manager Express (CUCME) Cisco Unified Communications Manager (CUCM) Session Initiation Protocol (SIP) Components Used The information in this document is based on these software and hardware versions: Cisco IOS Release 15.1(4)M4 on Cisco Integrated Services Routers (ISR): Series 2800, 3800, 2900, 3900 Cisco IOS Release 15.1(3)S4 on Cisco ASR 1000 Series Aggregation Services Routers Note: This configuration example is not limited to the software versions and hardware platforms listed above; this configuration also works with Cisco IOS Release 12.4(24)T5 on Cisco AS5400XM Universal Gateway.

Background Information SIP reliable provisional response was introduced in order to better integrate with a public switched telephone network (PSTN). The most common scenario is to establish the voice/audio path before completion of the call; therefore, the caller hears the announcement or music generated by the PSTN. For example, in below topology, the IP phone calls a PSTN conference bridge or some toll free numbers, and the callee plays a prompt before it answers the call. If the CUCM initiates the call with a delay offer (INVITE does not contain Session Description Protocol (SDP)), the caller will not hear the prompt. In other cases, the PSTN side generates a ringback tone. If media is not cut through before the call connects, the caller might not hear the ringback tone. SIP reliable provisional response can be used to resolve the above issue without involving extra media resources (such as Media Transfer Protocol (MTP)), as these provisional responses and PRACK messages provide additional opportunities for offer/answer exchanges. CUBE Configuration By default, CUBE supports reliable response with this configuration: voice service voip sip rel1xx supported 100rel This means, as a User Agent Client (UAC), if it receives 180/183 messages with header Require: 100rel, it will respond with PRACK; however, as a User Agent Server (UAS), it will not send out 180/183 with the header Require: 100rel. In order to force CUBE to send 18X with Require: 100rel (so that it will wait for PRACK from UAC), here is the configuration example: Global level: voice service voip sip rel1xx require 100rel Dial peer level: dial peer voice 1000 voip voice class sip rel1xx require 100rel

Note: The dial peer setting takes precedence over the global setting. CUCM Configuration By default, CUCM does not support reliable response. However, you can change the SIP trunk profile in order to configure it: 1. In the CUCM Administration interface, go to Device > Device Setting > SIP Profile. 2. Open the SIP profile used by a given SIP trunk. 3. Choose Send PRACK for all 1xx Messages from the SIP Rel1XX Options drop down list. 4. Reset the SIP trunk profile for the given SIP trunk. Note: If the given SIP trunk uses the default SIP trunk profile (Standard SIP Profile), it is best to copy to a new profile and apply to the SIP trunk; otherwise, the default SIP trunk profile will affect all SIP trunks. Note: Even if you make the above change, CUCM can support reliable responses only by sending PRACK as a UAC; however, for now, it cannot send 180/183 with the Require: 100rel header as a UAS. Typical SIP Messages If reliable repsonse is configured in the incoming dial peer on CUBE, a typical call will be similar to this: // CUBE receives INVITE with delay offer from CUCM. INVITE sip:2002@10.66.75.246:5060 SIP/2.0 Date: Thu, 04 Apr 2013 05:30:27 GMT Call Info: <sip:10.66.75.171:5060>;method="notify;event=telephone event;duration=500" Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow Events: presence, kpml P Asserted Identity: <sip:4832@10.66.75.171> Supported: 100rel,timer,resource priority,replaces,x cisco srtp fallback,geolocation Min SE: 7200 Cisco Guid: 3228672256 0000065536 0000000027 2873836042 Remote Party ID: <sip:4832@10.66.75.171>;party=calling;screen=yes;privacy=off User Agent: Cisco CUCM8.6 To: <sip:2002@10.66.75.246> Contact: <sip:4832@10.66.75.171:5060;transport=tcp> Expires: 180 Session Expires: 7200 Max Forwards: 70

SIP/2.0 100 Trying To: <sip:2002@10.66.75.246> Allow Events: telephone event // CUBE responds 183 with SDP which also contains Require: 100rel. SIP/2.0 183 Session Progress Require: 100rel RSeq: 3344 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow Events: telephone event Contact: <sip:2002@10.66.75.246:5060;transport=tcp> Supported: sdp anat Supported: X cisco srtp fallback Content Type: application/sdp Content Disposition: session;handling=required Content Length: 330 v=0 o=ciscosystemssip GW UserAgent 4874 2535 IN IP4 10.66.75.246 s=sip Call c=in IP4 10.66.75.246 t=0 0 m=audio 16442 RTP/AVP 8 0 18 101 19 c=in IP4 10.66.75.246 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone event/8000 a=fmtp:101 0 15 a=rtpmap:19 CN/8000 // CUBE receives PRACK from CUCM with SDP PRACK sip:2002@10.66.75.246:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.66.75.171:5060;branch=z9hG4bK246da4c33fa3e Date: Thu, 04 Apr 2013 05:30:27 GMT CSeq: 102 PRACKRAck: 3344 101 INVITE Allow Events: presence, kpml Max Forwards: 70 Content Type: application/sdp Content Length: 213 v=0 o=ciscosystemsccm SIP 169850 1 IN IP4 10.66.75.171 s=sip Call c=in IP4 10.66.75.89

t=0 0 m=audio 26662 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone event/8000 a=fmtp:101 0 15 // CUBE acknowledges the PRACK. SIP/2.0 200 OK Via: SIP/2.0/TCP 10.66.75.171:5060;branch=z9hG4bK246da4c33fa3e CSeq: 102 PRACK // The call is not anwered until now; however, calling and called parties have exchanged SDP, // and media path is established. // Other messages omitted. Troubleshooting In order to troubleshoot this issue on CUBE, these debugs must be enabled: debug voip ccapi inout debug ccsip message Symptom 1: CUBE sends out 180/183 without the Require: 100rel header. Verify that rel1xx require 100rel is configured under the approprate dial peer or voice service voip. Symptom 2: CUBE continues to send 180/183 with the Require: 100rel header to CUCM. This issue usually occurs when CUCM does not support reliable response. In order to resolve this issue, enable Rel1xx on CUCM. Related Information Technical Support & Documentation Cisco Systems Updated: May 16, 2013 Document ID: 116086