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Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

Last Updated: August 04, 2011

First Published: July 22, 2011

Last Updated: July 22, 2011

The Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature helps to avoid inter-operatibility issues by consuming the mid-call Re-INVITEs/UPDATEs from Cisco Unified Border Element.

Restrictions for Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

  • The Cisco UBE Mid-call Re-INVITE consumption feature works only in Cisco UBE and Session Initiation Protocol - Session Initiation Protocol (SIP) calls and not for SIP-H.323 or in TDM Gateways.
  • Cisco UBE Session Description Protocol (SDP) -passthrough does not work with Cisco UBE Mid-call Re-INVITE consumption.
  • When codec T is configured, the offer from Cisco UBE will only have audio codecs, and not video codecs and so the video codecs are not consumed.
  • Cisco UBE Mid-call Re-INVITE/UPDATE consumption works only if the flow-throughcommand is configured. If the flow-around command is configured, Re-INVITEs/UPDATEs will not be consumed.
  • Media Antitrombones are media loops in a SIP entity due to call transfer or call forward. When Media Antitrombone feature is configured, Cisco UBE moves dynamically to flow-aroundas a result media antithrombone takes precedence over the Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature.
  • Video transcoding is not supported when Mid-call Re-INVITE consumption is configured.
  • Secure Real-time Protocol - Real-time Protocol (SRTP-RTP) supplementary services and mid-call signaling are configured, so the Re-INVITEs/UPDATEs are not consumed.
  • Multicast Music On Hold (MMOH) is not supported and the Re-INVITEs/UPDATEs are passed over.
  • When the midcall-signaling passthru media-changecommand is configured and high-density transcoder is enabled, there might be some impact on Digital Signal Processing (DSP) resources as the transcoder might be used for all the calls.
  • Session timer is handled leg by leg whenever this feature is configured.

Information About Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

To configure Cisco UBE Mid-call Re-INVITE you should understand the following concepts:

Types of Mid-call Re-INVITE/UPDATE Consumption

Mid-call Re-INVITEs/UPDATEs consumption can be done in the following two ways:

Passthrough Media Change

Passthrough media change method optmizes or consumes mid-call media-related signaling within the call. Mid-call signaling changes will be passed through only when bidirectional media like T.38 or video is added. The command mid-call signaling passthru media-changeneeds to be configured to enable passthrough media change.

Block

The Block method blocks all mid-call media-related signaling to the specific SIP trunk. The command mid-call signaling block needs to be configured to enable this behavior. Video escalation and T.38 callflow are rejected when the mid-call signaling block command is configured. This command should be configured only when basic call is the focus and mid-call can be consumed.

Behavior of Mid-call Re-INVITE/UPDATE Consumption

The purpose of Mid-call Re-INVITE Consumption is to ensure smooth interoperatibility of supplementary services like audio Hold/Resume and call transfer. This feature should be used as a last resort only when there is no other option in Cisco UBE. This is because configuring this feature can break video-related features. For Delay-offer Re-INVITE, the configured codec will be passed as an offer in 200 message to change the codec, the transcoder is added in the answer.

How to Configure Cisco UBE Mid-call Re-INVITE Consumption

Configuring Passthrough SIP Messages at the Global Level

Perform this task to configure passthrough SIP messages d at the global level:

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    sip

5.    mid-call signaling passthru media-change

6.    exit


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
voice service voip


Example:

Router(config)# voice service voip

 

Enters voice service VOIP configuration mode.

 
Step 4
sip


Example:

Router(conf-voi-serv)# sip

 

Enters voice service VOIP SIP configuration mode.

 
Step 5
mid-call signaling passthru media-change


Example:

Router(conf-serv-sip)# mid-call signaling passthru media-change



Example:

Passes through SIP messages that involve media change.

 
 
Step 6
exit


Example:

Router(conf-serv-sip)# exit

 

Exits the current mode.

 

Configuring Passthrough SIP Messages at Dial Peer Level

Perform this task to configure passthrough SIP messages at the dial peer level. You need to perform this task at the dial peer level to consume all media-related mid-call Re-INVITEs/UPDATEs.


Note


If the Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature is configured on global and dial-peer level, dial-peer level takes precedence.
SUMMARY STEPS

1.    enable

2.    configure terminal

3.    dial-peer voice dial-peer tag voip

4.    mid-call signaling passthru media-change

5.    exit


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
dial-peer voice dial-peer tag voip


Example:

Router(config)# dial-peer voice 2 voip



Example:

Enter dial-peer voice configuration mode.

 
 
Step 4
mid-call signaling passthru media-change


Example:

Router(conf-dial-peer)# mid-call signaling passthru media-change



Example:

Passes through SIP messages that involve media change.

 
 
Step 5
exit


Example:

Router(config-dial-peer)# exit

 

Exits the current mode.

 

Blocking SIP Messages at the Global Level

Perform this task to block SIP messages at the global level:

SUMMARY STEPS

1.    enable

2.    configure terminal

3.    voice service voip

4.    sip

5.    mid-call signaling block

6.    exit


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
voice service voip


Example:

Router(config)# voice service voip

 

Enters voice service VOIP configuration mode.

 
Step 4
sip


Example:

Router(conf-voi-serv)# sip

 

Enters voice service VOIP SIP configuration mode.

 
Step 5
mid-call signaling block


Example:

Router(conf-serv-sip)# mid-call signaling block



Example:

Blocks all SIP messages during mid-call.

 
 
Step 6
exit


Example:

Router(conf-serv-sip)# exit

 

Exits the current mode.

 

Blocking SIP Messages at Dial Peer Level

Perform this task to block SIP messages at dial peer level.


Note


If the Cisco UBE Mid-call Re-INVITE/UPDATE consumption feature is configured on global and dial-peer level, dial-peer levels takes precedence.
SUMMARY STEPS

1.    enable

2.    configure terminal

3.    dial-peer voice dial-peer tag voip

4.    mid-call signaling block

5.    exit


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
dial-peer voice dial-peer tag voip


Example:

Router(config)# dial-peer voice 2 voip



Example:

Enter dial-peer voice configuration mode.

 
 
Step 4
mid-call signaling block


Example:

Router(conf-dial-peer)# mid-call signaling block

 

Blocks all SIP messages during mid-call.

 
Step 5
exit


Example:

Router(config-dial-peer)# exit

 

Exits the current mode.

 

Verifying and Troubleshooting Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

Perform this task to verify mid-call Re-INVITE/UPDATE routing support on Cisco UBE

SUMMARY STEPS

1.    enable

2.    debug ccsip all

3.    debug voip ccapi inout


DETAILED STEPS
Step 1   enable

Enables privileged EXEC mode.



Example:

Router> enable

Step 2   debug ccsip all

Use this command to enable SIP related debugging.



Example:

Router# debug ccsip all Received: INVITE sip:5555555555@[2208:1:1:1:1:1:1:1118]:5060 SIP/2.0 Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1115]:5060;branch=z9hG4bK83AE3 Remote-Party-ID: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;party=calling;screen=no;privacy=off From: <sip:2222222222@[2208:1:1:1:1:1:1:1115]>;tag=627460F0-1259 To: <sip:5555555555@[2208:1:1:1:1:1:1:1118]> Date: Tue, 01 Mar 2011 08:49:48 GMT Call-ID: B30FCDEB-431711E0-8EDECB51-E9F6B1F1@2208:1:1:1:1:1:1:1115 Supported: 100rel,timer,resource-priority,replaces Require: sdp-anat Min-SE: 1800 Cisco-Guid: 2948477781-1125585376-2396638033-3925258737 User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1298969388 Contact: <sip:2222222222@[2208:1:1:1:1:1:1:1115]:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 495 v=0 o=CiscoSystemsSIP-GW-UserAgent 7880 7375 IN IP6 2208:1:1:1:1:1:1:1115 s=SIP Call c=IN IP6 2208:1:1:1:1:1:1:1115 t=0 0 a=group:ANAT 1 2 m=audio 17836 RTP/AVP 0 101 19 c=IN IP6 2208:1:1:1:1:1:1:1115 a=mid:1 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 m=audio 18938 RTP/AVP 0 101 19 c=IN IP4 9.45.36.111 a=mid:2 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 “Received: INVITE sip:2222222222@[2208:1:1:1:1:1:1:1117]:5060 SIP/2.0 Via: SIP/2.0/UDP [2208:1:1:1:1:1:1:1116]:5060;branch=z9hG4bK38ACE Remote-Party-ID: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;party=calling;screen=no;privacy=off From: <sip:5555555555@[2208:1:1:1:1:1:1:1116]>;tag=4FE8C9C-1630 To: <sip:2222222222@[2208:1:1:1:1:1:1:1117]>;tag=1001045C-992 Date: Thu, 10 Feb 2011 12:15:08 GMT Call-ID: 5DEDB77E-ADC11208-808BE770-8FCACF34@2208:1:1:1:1:1:1:1117 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 1432849350-0876876256-2424621905-3925258737 User-Agent: Cisco-SIPGateway/IOS-15.1(3.14.2)PIA16 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1297340108 Contact: <sip:5555555555@[2208:1:1:1:1:1:1:1116]:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 424 v=0 o=CiscoSystemsSIP-GW-UserAgent 8002 7261 IN IP6 2208:1:1:1:1:1:1:1116 s=SIP Call c=IN IP6 2208:1:1:1:1:1:1:1116 t=0 0 m=image 17278 udptl t38 c=IN IP6 2208:1:1:1:1:1:1:1116 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:320 a=T38FaxUdpEC:t38UDPRedundancy

Step 3   debug voip ccapi inout


Example:

The voice gateway sets up negotiating capability with the terminating VoIP leg. *Mar 1 15:36:05.028: //45/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind: (dstVdbPtr=0x637E C1E0, dstCallId=0x2C, srcCallId=0x2D, caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0 codec_bytes=20, signal_type=2}) *Mar 1 15:36:05.028: //45/xxxxxxxxxxxx/CCAPI/ cc_api_caps_ind: (Playout: mode 0, initial 60,min 40, max 300)

The capabilities are acknowledged for both call legs.



Example:

*Mar 1 15:36:05.028: //45/xxxxxxxxxxxx/CCAPI/cc_api_caps_ack: (dstVdbPtr=0x637E C1E0, dstCallId=0x2C, srcCallId=0x2D, caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0 codec_bytes=20, signal_type=2, seq_num_start=2944}) *Mar 1 15:36:05.028: //44/xxxxxxxxxxxx/CCAPI/cc_api_caps_ack: (dstVdbPtr=0x62EC 61A4, dstCallId=0x2D, srcCallId=0x2C, caps={codec=0x4, fax_rate=0x2, vad=0x2, modem=0x0 codec_bytes=20, signal_type=2, seq_num_start=2944}) *Mar 1 15:36:05.032: //44/xxxxxxxxxxxx/CCAPI/cc_api_voice_mode_event: callID=0x2C *Mar 1 15:36:05.032: //44/45F2AAE28044/CCAPI/cc_api_voice_mode_event: Call Pointer =634A430C *Mar 1 15:36:05.032: //44/xxxxxxxxxxxx/SSAPP:-1:-1/sess_appl: ev(52=CC_EV_VOICE _MODE_DONE), cid(44), disp(0) *Mar 1 15:36:05.032: //44/45F2AAE28044/SSAPP:10002:21/ssaTraceSct: Router# Router# cid(44)st(SSA_CS_ACTIVE)ev(SSA_EV_VOICE_MODE_DONE) oldst(SSA_CS_CONFERENCING)cfid(21)csize(2)in(1)fDest(1) *Mar 1 15:36:05.032: //44/45F2AAE28044/SSAPP:10002:21/ssaTraceSct: -cid2(45)st2 (SSA_CS_ACTIVE)oldst2(SSA_CS_ALERT_RCVD) *Mar 1 15:36:05.032: //44/45F2AAE28044/SSAPP:10002:21/ssaIgnore: cid(44), st(SS A_CS_ACTIVE),oldst(5), ev(52) Router# Router#! digit punched Router#

The phone at the terminating gateway enters digit1.



Example:

*Mar 1 15:36:11.204: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin: (dstVdbPtr=0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D, digit=1, digit_begin_flags=0x0, rtp_timestamp=0x0 rtp_expiration=0x0, dest_mask=0x2) *Mar 1 15:36:11.504: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end: (dstVdbPtr= 0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D, digit=1,duration=300,xruleCallingTag=0,xruleCalledTag=0, dest_mask=0x2), digit_tone_mode=0

The phone at the terminating gateway enters digit 2.



Example:

*Mar 1 15:36:11.604: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin: (dstVdbPt r=0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D, digit=2, digit_begin_flags=0x0, rtp_timestamp=0x0 rtp_expiration=0x0, dest_mask=0x2) *Mar 1 15:36:11.904: //45/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end: (dstVdbPtr= 0x637EC1E0, dstCallId=0x2C, srcCallId=0x2D, digit=2,duration=300,xruleCallingTag=0,xruleCalledTag=0, dest_mask=0x2), digit_tone_mode=0 Router# Router# *Mar 1 15:36:14.476: //-1/xxxxxxxxxxxx/CCAPI/cc_handle_periodic_timer: Calling the callback, ccTimerctx - 0x628B6330 *Mar 1 15:36:14.476: //-1/xxxxxxxxxxxx/CCAPI/ccTimerStart: ccTimerctx - 0x628B6330 Router# Router# !call hung up The user at the terminating gateway hangs up the call. Router#


Configuration Examples for Cisco UBE Mid-call Re-INVITE Consumption

Example Configuring Passthrough SIP Messages at the Global Level

The following example shows how to passthrough SIP messages at the global level:

Router(config)# voice service voip
Router(config-voi-serv)# no ip address trusted authenticate
Router(config-voi-serv)# allow-connections sip to sip
Router(config-voi-serv)# sip
Router(config-serv-sip)# midcall-signaling passthru media-change

Example Configuring Passthrough SIP Messages at Dial Peer Level

The following example shows how to passthrough SIP messages at the dial peer Level:

dial-peer voice 600 voip
 destination-pattern 2222222222
 session protocol sipv2
 session target ipv4:9.45.38.39:9001
voice-class sip midcall-signaling passthru media-change
 incoming called-number 1111111111
voice-class codec  2 offer-all
dial-peer voice 400 voip
 destination-pattern 1111111111
 session protocol sipv2
 session target ipv4:9.45.38.39:9000
 incoming called-number 2222222222
voice-class codec 1 offer-all

Example: Blocking SIP Messages at the Global Level

The following example shows how to block SIP messages at the global Level

Router(config)#voice service voip
Router(config-voi-serv)#no ip address trusted authenticate
Router(config-voi-serv)#allow-connections sip to sip
Router(config-voi-serv)#sip
Router(config-serv-sip)#midcall-signaling block

Example Blocking SIP Messages at Dial Peer Level

dial-peer voice 107 voip
 destination-pattern 74000
 session protocol sipv2
 session target ipv4:9.45.36.9
 incoming called-number 84000
 voice-class codec 1 offer-all
!
dial-peer voice 110 voip
 destination-pattern 84000
 session protocol sipv2
 session target ipv4:9.45.35.2
 incoming called-number 74000
 voice-class codec 1 offer-all
 voice-class sip midcall-signaling block
!

Additional References

Related Documents

Related Topic

Document Title

Cisco IOS commands

Cisco IOS Master Commands List, All Releases

Cisco UBE fundaments and basic setup

Cisco Unified Border Element Fundaments and Basic Setup

Cisco IOS voice configuration

Cisco IOS Voice Configuration Library

Cisco IOS voice commands

Cisco IOS Voice Command Reference

Cisco IOS debug commands

Cisco IOS Debug Command Reference

Standards

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To locate and download MIBs for selected platforms, Cisco software releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://www.cisco.com/go/mibs

RFCs

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Technical Assistance

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Link

The Cisco Support and Documentation website provides online resources to download documentation, software, and tools. Use these resources to install and configure the software and to troubleshoot and resolve technical issues with Cisco products and technologies. Access to most tools on the Cisco Support and Documentation website requires a Cisco.com user ID and password.

http://www.cisco.com/cisco/web/support/index.html

Feature Information for Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

GUID-49C47880-20C3-40B5-A513-C1A80291D0918 lists the release history for this feature.

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn . An account on Cisco.com is not required.


Note


GUID-49C47880-20C3-40B5-A513-C1A80291D0918 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.
Table 1Feature Information for Cisco UBE Mid-call Re-INVITE /UPDATE Consumption

Feature Name

Releases

Feature Information

Cisco UBE Mid-call Re-INVITE/UPDATE Consumption

15.2(1)T

The Cisco UBE Mid-call Re-INVITE consumption helps to avoid inter-operatibility issues by consuming the mid-call Re-INVITEs from Cisco UBE.

The following section provides information about this feature:

The following commands were introduced or modified: mid-call signaling.

Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental. © 2011 Cisco Systems, Inc. All rights reserved