Table Of Contents
Enhancements to the Session Initiation Protocol for VoIP on Cisco Access Platforms
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring SIP Support for VoIP Dial Peers
Changing the Configuration of the SIP User Agent (UA)
Configuring Phone Number Translation Rules
Verifying the SIP Feature Configuration
Basic SIP Configuration Example
Enhancements to the Session Initiation Protocol for VoIP on Cisco Access Platforms
Document Update Alert
This document was originally produced for Cisco IOS Release 12.2(11)T. This feature has been updated in subsequent releases, and more recent documentation is available.
If you are using Cisco IOS Release 12.2(11)T or higher, refer to the following documentation in the Cisco IOS Voice Configuration Library, Release 12.3:
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Cisco IOS SIP Configuration Guide
Feature History
This document describes the enhancements to the Session Initiation Protocol (SIP) for Voice over Internet Protocol (VoIP) on Cisco access platforms in Cisco IOS Release 12.2(11)T.
This document includes the following sections:
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Feature Overview
VoIP currently implements the ITU H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. The Session Initiation Protocol (SIP) is a new protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999.
The Cisco SIP functionality enables Cisco access platforms to signal the setup of voice and multimedia calls over IP networks. The SIP feature also provides non-proprietary advantages in the areas of:
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The SIP feature enhancements include the following:
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Table 1 lists the PSTN cause codes that can be sent as an ISDN cause information element (IE) and the corresponding SIP event for each.
Table 1 PSTN Cause Code to SIP Event Mappings
Table 2 lists the SIP events and the corresponding PSTN cause codes for each.
Table 2 SIP Event to PSTN Cause Code Mapping
Benefits
The SIP feature enhancements enable SIP gateways to do the following:
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Restrictions
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Related Features and Technologies
The SIP feature is dependent upon the interoperability of Service Provider Features for VoIP.
Related Documents
The following documents contain information related to the Cisco SIP functionality:
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Supported Platforms
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Determining Platform Support Through Cisco Feature Navigator
Cisco IOS software is packaged in feature sets that are supported on specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.
Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.
To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:
Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
Availability of Cisco IOS Software Images
Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.
Supported Standards, MIBs, and RFCs
Standards
No new or modified standards are supported by this feature.
MIBs
No new or modified MIBs are supported by this feature.
To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL:
http://tools.cisco.com/ITDIT/MIBS/servlet/index
If Cisco MIB Locator does not support the MIB information that you need, you can also obtain a list of supported MIBs and download MIBs from the Cisco MIBs page at the following URL:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml
To access Cisco MIB Locator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:
RFCs
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Prerequisites
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For more information about configuring IP, refer to Cisco IOS IP and IP Routing Configuration Guide.
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Configuration Tasks
See the following sections for configuration tasks for this feature. Each task in the list is identified as either required or optional.
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Configuring SIP Support for VoIP Dial Peers
To configure SIP support for a VoIP dial peer, you must enter the following commands beginning in global configuration mode:
Changing the Configuration of the SIP User Agent (UA)
It is not necessary to configure a SIP UA to place a call. A SIP UA is configured to listen by default. However, if you want to adjust any of the settings, you can do so by using the following commands beginning in global configuration mode:
Step 1
Router(config)# sip-ua
Enters SIP user agent (sip-ua) mode to configure SIP-UA related commands.
Step 2
Router(config-sip-ua)# transport {udp|tcp}
Configures the SIP user agent (sip-ua) for SIP signaling messages. The default value is udp.
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Router(config-sip-ua)# sip-server {dns:[host-name]|ipv4:ip_address}
Enters the host name or IP address of the SIP server interface.
Step 4
Router(config-sip-ua)# timers trying number
Sets time to wait for a response.
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Router(config-sip-ua)# timers expires time
Limits the time duration (in milliseconds) of a search for an INVITE.
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Router(config-sip-ua)# retry invite number
Configures the SIP signaling timers for retry attempts.
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Router(config-sip-ua)# max-forwards number_of_hops
Limits the number of proxy or redirect servers that can forward a request.
Configuring SIP Call Transfer
The following example illustrates how to configure call transfer. In Figure 1, User A and User C are in an established call. User C then transfers the call to User B. This results in a call being established between User A and User B. User C is then disconnected with User A, regardless of whether the transfer fails or succeeds.
When a call originates or terminates on a gateway, either the calling party number, the called party number, or the port is used (depending on the scenario) to match a dial peer to determine the basic call characteristics. One of the characteristics to determine is which application to use for the call. For the call transfer to succeed, the matching dial peer must have application set to "session" on the gateway that is controlling the transfer. (This is the gateway that receives the Bye with an Also header).
There are two scenarios for dial-peer matching based on whether the call is coming from a POTS interface or from the IP network.
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Figure 1 Call Transfer Example
In this example, Gateway 1 handles the transfer (recipient of the Bye with the Also header). User C invokes the transfer service (originator of the Bye with the Also header). There are two scenarios in which a dial peer match must have application set to "session" for the transfer to succeed:
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The matching dial peer must have application set to "session" if transfer is later invoked by User C.
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To configure SIP call transfer for a POTS dial peer, enter the following commands beginning in global configuration mode:
To configure SIP call transfer for a VoIP dial peer, enter the following commands beginning in global configuration mode.
To configure a POTS dial peer with the session application, enter the following commands beginning in config-dial-peer configuration mode:
To configure a VoIP dial peer with a destination pattern, enter the following commands beginning in config-dial-peer configuration mode:
To configure a VoIP dial peer with an incoming called-number, enter the following commands beginning in config-dial-peer configuration mode:
To configure a VoIP dial peer with an incoming called-number, enter the following commands beginning in config-dial-peer configuration mode:
Configuring Phone Number Translation Rules
By default, the SIP gateway tags called numbers that have 11 or more digits as "international" when sending SETUP messages to the PSTN switch. In some cases, such as situations where the user must dial 9 to access an outside line, this assumption may not be correct.
To accommodate such situations, you can define translation rules on the outbound POTS dial peer to convert the "type of number" to the correct value. Translation rules manipulate the called number digits and the "type of number" value associated with the called digits.
To define translation rules on a POTS dial peer, enter the following commands beginning in global configuration mode:
For more information about the commands used to configure translation rules, see the
Dial Peer Enhancements documentation on Cisco.com.Verifying the SIP Feature Configuration
Enter the following show commands to verify your configuration:
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Troubleshooting Tips
Use the following debug commands to troubleshoot your configuration:
Configuration Examples
This section contains examples of the following:
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Basic SIP Configuration Example
The following shows an example of the output that appears when you enter the show running configuration command.
Router# show running configurationBuilding configuration...Current configuration:!version 12.1service timestamps debug datetimeservice timestamps log uptimeno service password-encryption!hostname Router1!!!clock timezone GMT 5voice-card 1!ip subnet-zeroip tcp path-mtu-discoveryip name-server 172.18.192.48!isdn voice-call-failure 0!!controller T1 1/0framing esfclock source line primarylinecode b8zs!controller T1 1/1!!voice-port 2/0/0!voice-port 2/0/1!voice class codec 1codec preference 1 g711alawcodec preference 2 g723r63codec preference 3 g723r53!!dial-peer voice 100 potsdestination-pattern 3660110port 2/0/0!dial-peer voice 200 potsapplication sessiondestination-pattern 3660120port 2/0/1!dial-peer voice 101 voipdestination-pattern 3660210session protocol sipv2session target ipv4:166.34.244.73codec g711ulaw!dial-peer voice 201 voipapplication sesiondestination-pattern 3660220session protocol sipv2session target dns:3660-2.sip.comcodec g711ulaw!dial-peer voice 999 voipdestination-pattern 5551111session protocol sipv2session target ipv4:161.44.53.89session transport tcp!dial-peer voice 300 potsdestination-pattern 2101100!dial-peer voice 350 voipdestination-pattern 3100607session protocol sipv2session target ipv4:172.18.192.197codec g711ulaw!dial-peer voice 301 voipapplication sessiondestination-pattern 1234session protocol sipv2session target ipv4:172.18.192.193codec g711ulaw!dial-peer voice 333 voipapplication sessiondestination-pattern 1235session protocol sipv2session target ipv4:172.18.192.199codec g711ulaw!dial-peer voice 888 voipdestination-pattern 888session protocol sipv2session target ipv4:161.44.53.89session transport tcpcodec g711ulaw!dial-peer voice 260011 voipdestination-pattern 260011session protocol sipv2session target ipv4:172.18.192.164codec g711ulaw!dial-peer voice 444 voipdestination-pattern 2339000session protocol sipv2session target ipv4:172.18.192.205codec g711ulaw!dial-peer voice 111 voipdestination-pattern 111session protocol sipv2session target sip-servercodec g711ulaw!dial-peer voice 7777777 voipdestination-pattern 19197777777session protocol sipv2session target ipv4:172.18.192.38codec g711ulaw!!sip-uaretry invite 2retry response 2retry bye 2retry cancel 2no inband-alertingsip-server dns:!!interface FastEthernet0/0ip address 172.18.192.194 255.255.255.0load-interval 30speed autohalf-duplex!interface FastEthernet0/1ip address 166.34.245.230 255.255.255.224load-interval 30speed autohalf-duplex!ip classlessip route 0.0.0.0 0.0.0.0 172.18.192.1ip route 166.34.0.0 255.255.0.0 166.34.245.225no ip http server!access-list 101 permit ip host 10.0.2.30 host 10.0.2.31access-list 101 deny udp any eq rip anyaccess-list 101 deny udp any any eq ripaccess-list 101 deny udp any eq isakmp anyaccess-list 101 deny udp any any eq isakmpaccess-list 101 permit ip any anysnmp-server engineID local 000000090200003094202740snmp-server community public RW!line con 0exec-timeout 0 0transport input noneline aux 0line vty 0 4password xxxlogin!endTranslation Rule Example
The following example illustrates a translation rule for dialing national numbers in the situation where the user must dial 9 to access an outside line. In the rule command in this example:
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The result of this command is that any outgoing call that is destined for a number that starts with 91 and that is considered by the gateway to be an international number, will be sent to the PSTN as a national number with a prefix of 1.
translation-rule 10Rule 1 91% 1 international national!!!dial-peer voice 10 potsdestination-pattern 91..........translate-outgoing called 10port 1:D!The following example illustrates a translation rule for dialing national numbers in the situation where the user does not need to dial 9 to access an outside line.
translation-rule 10Rule 1 1% 1 international national!!!dial-peer voice 10 potsdestination-pattern 1..........translate-outgoing called 10port 1:Dprefix 1!The following example illustrates a translation rule for dialing international numbers in the situation where the user must dial 9 to access an outside line.
translation-rule 20Rule 1 9011% 011 unknown international!!!dial-peer voice 10 potsdestination-pattern 9011Ttranslate-outgoing called 20port 1:D!The following example illustrates a translation rule for dialing international numbers in the situation where the user does not need to dial 9 to access an outside line.
translation-rule 20Rule 1 011% 011 unknown international!!!dial-peer voice 10 potsdestination-pattern 011Ttranslate-outgoing called 20port 1:Dprefix 011!Call Transfer Example
The following example shows how to configure SIP call transfer for a VoIP dial peer:
Router(config)# dial-peer voice number voip
Router(config-dial-peer)# application session
Router(config-dial-peer)# destination-
pattern patternRouter(config-dial-peer)# session target ipv4:x.x.x.x
The following example shows how to configure SIP call transfer for a VoIP dial peer:
Router(config)# dial-peer voice number voip
Router(config-dial-peer)# application session
Router(config-dial-peer)# destination-
pattern patternRouter(config-dial-peer)# session target ipv4:x.x.x.x
Command Reference
This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release12.3T command reference publications.
This section documents the following commands:
debug ccsip all
To enable all SIP-related debugging, enter the debug ccsip all command in privileged EXEC configuration mode. To disable debugging output, use the no form of this command.
debug ccsip all
no debug ccsip all
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
The debug ccsip all command enables the following SIP debug commands:
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Examples
From one side of the call, the debug output is as follows:
Router1# debug ccsip allAll SIP call tracing enabledRouter1#*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)*Mar 6 14:10:42: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_idle_call_setup*Mar 6 14:10:42: act_idle_call_setup:Not using Voice Class Codec*Mar 6 14:10:42: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 160*Mar 6 14:10:42: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING)*Mar 6 14:10:42: REQUEST CONNECTION TO IP:166.34.245.231 PORT:5060*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING)*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_idle_connection_created*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 166.34.245.231:5060, local_port 54113*Mar 6 14:10:42: sipSPIAddLocalContact*Mar 6 14:10:42: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sip_stats_method*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE)*Mar 6 14:10:42: Sent:INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Sat, 06 Mar 2002 19:10:42 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Cisco-Guid: 2881152943-2184249548-0-483039712User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEMax-Forwards: 6Timestamp: 731427042Contact: <sip:3660110@166.34.245.230:5060;user=phone>Expires: 180Content-Type: application/sdpContent-Length: 137v=0o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20208 RTP/AVP 0*Mar 6 14:10:42: Received:SIP/2.0 100 TryingVia: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:36:40 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Timestamp: 731427042Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Length: 0*Mar 6 14:10:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_sentinvite_new_message*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:10:42: Roundtrip delay 4 milliseconds for method INVITE*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)*Mar 6 14:10:42: Received:SIP/2.0 180 RingingVia: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:36:40 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Timestamp: 731427042Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Type: application/sdpContent-Length: 137v=0o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20038 RTP/AVP 0*Mar 6 14:10:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_recdproc_new_message*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:10:42: Roundtrip delay 8 milliseconds for method INVITE*Mar 6 14:10:42: HandleSIP1xxRinging: SDP MediaTypes negotiation successful!Negotiated Codec : g711ulaw , bytes :160Inband Alerting : 0*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING)*Mar 6 14:10:46: Received:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FDate: Mon, 08 Mar 2002 22:36:40 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Timestamp: 731427042Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledContact: <sip:3660210@166.34.245.231:5060;user=phone>CSeq: 101 INVITEContent-Type: application/sdpContent-Length: 137v=0o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20038 RTP/AVP 0*Mar 6 14:10:46: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060*Mar 6 14:10:46: CCSIP-SPI-CONTROL: act_recdproc_new_message*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:10:46: Roundtrip delay 3536 milliseconds for method INVITE*Mar 6 14:10:46: CCSIP-SPI-CONTROL: act_recdproc_new_message: SDP MediaTypes negotiation successful!Negotiated Codec : g711ulaw , bytes :160*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sipSPIReconnectConnection*Mar 6 14:10:46: Queued event from SIP SPI : SIPSPI_EV_RECONNECT_CONNECTION*Mar 6 14:10:46: CCSIP-SPI-CONTROL: recv_200_OK_for_invite*Mar 6 14:10:46: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sip_stats_method*Mar 6 14:10:46: 0x624CFEF8 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) to (STATE_ACTIVE, SUBSTATE_NONE)*Mar 6 14:10:46: The Call Setup Information is :Call Control Block (CCB) : 0x624CFEF8State of The Call : STATE_ACTIVETCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.230Source IP Port (Media): 20208Destn IP Address (Media): 166.34.245.231Destn IP Port (Media): 20038Destn SIP Addr (Control) : 166.34.245.231Destn SIP Port (Control) : 5060Destination Name : 166.34.245.231*Mar 6 14:10:46: HandleUdpReconnection: Udp socket connected for fd: 1 with 166.34.245.231:5060*Mar 6 14:10:46: Sent:ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FDate: Sat, 06 Mar 2002 19:10:42 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Max-Forwards: 6Content-Type: application/sdpContent-Length: 137CSeq: 101 ACKv=0o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20208 RTP/AVP 0*Mar 6 14:10:46: CCSIP-SPI-CONTROL: ccsip_caps_ind*Mar 6 14:10:46: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160*Mar 6 14:10:46: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE*Mar 6 14:10:46: CCSIP-SPI-CONTROL: ccsip_caps_ack*Mar 6 14:10:50: Received:BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.231:54835From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FTo: "3660110" <sip:3660110@166.34.245.230>Date: Mon, 08 Mar 2002 22:36:44 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledMax-Forwards: 6Timestamp: 731612207CSeq: 101 BYEContent-Length: 0*Mar 6 14:10:50: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:54835*Mar 6 14:10:50: CCSIP-SPI-CONTROL: act_active_new_message*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sact_active_new_message_request*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sip_stats_method*Mar 6 14:10:50: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(16) for outgoing call*Mar 6 14:10:50: 0x624CFEF8 : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)*Mar 6 14:10:50: Sent:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.231:54835From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FTo: "3660110" <sip:3660110@166.34.245.230>Date: Sat, 06 Mar 2002 19:10:50 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledTimestamp: 731612207Content-Length: 0CSeq: 101 BYE*Mar 6 14:10:50: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT*Mar 6 14:10:50: CCSIP-SPI-CONTROL: act_disconnecting_disconnect*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sipSPICallCleanup*Mar 6 14:10:50: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION*Mar 6 14:10:50: CLOSE CONNECTION TO CONNID:1*Mar 6 14:10:50: sipSPIIcpifUpdate :CallState: 4 Playout: 1755 DiscTime:48305031 ConnTime 48304651*Mar 6 14:10:50: 0x624CFEF8 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)*Mar 6 14:10:50: The Call Setup Information is :Call Control Block (CCB) : 0x624CFEF8State of The Call : STATE_DEADTCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.230Source IP Port (Media): 20208Destn IP Address (Media): 166.34.245.231Destn IP Port (Media): 20038Destn SIP Addr (Control) : 166.34.245.231Destn SIP Port (Control) : 5060Destination Name : 166.34.245.231*Mar 6 14:10:50:Disconnect Cause (CC) : 16Disconnect Cause (SIP) : 200*Mar 6 14:10:50: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060Router1#From the other side of the call, the debug output is as follows:
Router2# debug ccsip allAll SIP call tracing enabledRouter2#*Mar 8 17:36:40: Received:INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Sat, 06 Mar 2002 19:10:42 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Cisco-Guid: 2881152943-2184249548-0-483039712User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEMax-Forwards: 6Timestamp: 731427042Contact: <sip:3660110@166.34.245.230:5060;user=phone>Expires: 180Content-Type: application/sdpContent-Length: 137v=0o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20208 RTP/AVP 0*Mar 8 17:36:40: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:54113*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sipSPISipIncomingCall*Mar 8 17:36:40: 0x624D8CCC : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)*Mar 8 17:36:40: CCSIP-SPI-CONTROL: act_idle_new_message*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sact_idle_new_message_invite*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sip_stats_method*Mar 8 17:36:40: sact_idle_new_message_invite:Not Using Voice Class Codec*Mar 8 17:36:40: sact_idle_new_message_invite: Preferred codec[0] type: g711ulaw Bytes :160*Mar 8 17:36:40: sact_idle_new_message_invite: Media Negotiation successful for anincoming call*Mar 8 17:36:40: sact_idle_new_message_invite: Negotiated Codec : g711ulaw, bytes :160Preferred Codec : g711ulaw, bytes :160*Mar 8 17:36:40: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:36:40: Num of Contact Locations 1 3660110 166.34.245.230 5060*Mar 8 17:36:40: 0x624D8CCC : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_RECD_INVITE_CALL_SETUP)*Mar 8 17:36:40: Sent:SIP/2.0 100 TryingVia: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:36:40 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Timestamp: 731427042Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Length: 0*Mar 8 17:36:40: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING*Mar 8 17:36:40: CCSIP-SPI-CONTROL: act_recdinvite_proceeding*Mar 8 17:36:40: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_ALERTING*Mar 8 17:36:40: CCSIP-SPI-CONTROL: ccsip_caps_ind*Mar 8 17:36:40: ccsip_caps_ind: codec(negotiated) = 5(Bytes 160)*Mar 8 17:36:40: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160*Mar 8 17:36:40: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE*Mar 8 17:36:40: CCSIP-SPI-CONTROL: ccsip_caps_ack*Mar 8 17:36:40: CCSIP-SPI-CONTROL: act_recdinvite_alerting*Mar 8 17:36:40: 180 Ringing with SDP - not likely*Mar 8 17:36:40: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:36:40: 0x624D8CCC : State change from (STATE_RECD_INVITE, SUBSTATE_RECD_INVITE_CALL_SETUP) to (STATE_SENT_ALERTING, SUBSTATE_NONE)*Mar 8 17:36:40: Sent:SIP/2.0 180 RingingVia: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:36:40 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Timestamp: 731427042Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Type: application/sdpContent-Length: 137v=0o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20038 RTP/AVP 0*Mar 8 17:36:44: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_CONNECT*Mar 8 17:36:44: CCSIP-SPI-CONTROL: act_sentalert_connect*Mar 8 17:36:44: sipSPIAddLocalContact*Mar 8 17:36:44: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:36:44: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:36:44: 0x624D8CCC : State change from (STATE_SENT_ALERTING, SUBSTATE_NONE) to (STATE_SENT_SUCCESS, SUBSTATE_NONE)*Mar 8 17:36:44: Sent:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FDate: Mon, 08 Mar 2002 22:36:40 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Timestamp: 731427042Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledContact: <sip:3660210@166.34.245.231:5060;user=phone>CSeq: 101 INVITEContent-Type: application/sdpContent-Length: 137v=0o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20038 RTP/AVP 0*Mar 8 17:36:44: Received:ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:54113From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FDate: Sat, 06 Mar 2002 19:10:42 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Max-Forwards: 6Content-Type: application/sdpContent-Length: 137CSeq: 101 ACKv=0o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20208 RTP/AVP 0*Mar 8 17:36:44: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:54113*Mar 8 17:36:44: CCSIP-SPI-CONTROL: act_sentsucc_new_message*Mar 8 17:36:44: CCSIP-SPI-CONTROL: sip_stats_method*Mar 8 17:36:44: 0x624D8CCC : State change from (STATE_SENT_SUCCESS, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_NONE)*Mar 8 17:36:44: The Call Setup Information is :Call Control Block (CCB) : 0x624D8CCCState of The Call : STATE_ACTIVETCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.231Source IP Port (Media): 20038Destn IP Address (Media): 166.34.245.230Destn IP Port (Media): 20208Destn SIP Addr (Control) : 166.34.245.230Destn SIP Port (Control) : 5060Destination Name : 166.34.245.230*Mar 8 17:36:47: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT*Mar 8 17:36:47: CCSIP-SPI-CONTROL: act_active_disconnect*Mar 8 17:36:47: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_CONNECTING)*Mar 8 17:36:47: REQUEST CONNECTION TO IP:166.34.245.230 PORT:5060*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING) to (STATE_ACTIVE, SUBSTATE_CONNECTING)*Mar 8 17:36:47: CCSIP-SPI-CONTROL: act_active_connection_created*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection: Connid(1) created to 166.34.245.230:5060, local_port 54835*Mar 8 17:36:47: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sip_stats_method*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING) to (STATE_DISCONNECTING, SUBSTATE_NONE)*Mar 8 17:36:47: Sent:BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.231:54835From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FTo: "3660110" <sip:3660110@166.34.245.230>Date: Mon, 08 Mar 2002 22:36:44 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledMax-Forwards: 6Timestamp: 731612207CSeq: 101 BYEContent-Length: 0*Mar 8 17:36:47: Received:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.231:54835From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7FTo: "3660110" <sip:3660110@166.34.245.230>Date: Sat, 06 Mar 2002 19:10:50 GMTCall-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledTimestamp: 731612207Content-Length: 0CSeq: 101 BYE*Mar 8 17:36:47: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:54113*Mar 8 17:36:47: CCSIP-SPI-CONTROL: act_disconnecting_new_message*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:36:47: Roundtrip delay 4 milliseconds for method BYE*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICallCleanup*Mar 8 17:36:47: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION*Mar 8 17:36:47: CLOSE CONNECTION TO CONNID:1*Mar 8 17:36:47: sipSPIIcpifUpdate :CallState: 4 Playout: 1265 DiscTime:66820800 ConnTime 66820420*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)*Mar 8 17:36:47: The Call Setup Information is :Call Control Block (CCB) : 0x624D8CCCState of The Call : STATE_DEADTCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.231Source IP Port (Media): 20038Destn IP Address (Media): 166.34.245.230Destn IP Port (Media): 20208Destn SIP Addr (Control) : 166.34.245.230Destn SIP Port (Control) : 5060Destination Name : 166.34.245.230*Mar 8 17:36:47:Disconnect Cause (CC) : 16Disconnect Cause (SIP) : 200*Mar 8 17:36:47: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060Related Commands
debug ccsip calls
To show all SIP Service Provider Interface (SPI) call tracing, enter the debug ccsip calls command in privileged EXEC configuration mode. To disable debugging output, use the no form of this command.
debug ccsip calls
no debug ccsip calls
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command traces the SIP call details as they are updated in the SIP call control block.
Examples
From one side of the call, the debug output is as follows:
Router1# debug ccsip callsSIP Call statistics tracing is enabledRouter1#*Mar 6 14:12:33: The Call Setup Information is :Call Control Block (CCB) : 0x624D078CState of The Call : STATE_ACTIVETCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.230Source IP Port (Media): 20644Destn IP Address (Media): 166.34.245.231Destn IP Port (Media): 20500Destn SIP Addr (Control) : 166.34.245.231Destn SIP Port (Control) : 5060Destination Name : 166.34.245.231*Mar 6 14:12:40: The Call Setup Information is :Call Control Block (CCB) : 0x624D078CState of The Call : STATE_DEADTCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.230Source IP Port (Media): 20644Destn IP Address (Media): 166.34.245.231Destn IP Port (Media): 20500Destn SIP Addr (Control) : 166.34.245.231Destn SIP Port (Control) : 5060Destination Name : 166.34.245.231*Mar 6 14:12:40:Disconnect Cause (CC) : 16Disconnect Cause (SIP) : 200Router1#From the other side of the call, the debug output is as follows:
Router2# debug ccsip callsSIP Call statistics tracing is enabledRouter2#*Mar 8 17:38:31: The Call Setup Information is :Call Control Block (CCB) : 0x624D9560State of The Call : STATE_ACTIVETCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.231Source IP Port (Media): 20500Destn IP Address (Media): 166.34.245.230Destn IP Port (Media): 20644Destn SIP Addr (Control) : 166.34.245.230Destn SIP Port (Control) : 5060Destination Name : 166.34.245.230*Mar 8 17:38:38: The Call Setup Information is :Call Control Block (CCB) : 0x624D9560State of The Call : STATE_DEADTCP Sockets Used : NOCalling Number : 3660110Called Number : 3660210Negotiated Codec : g711ulawSource IP Address (Media): 166.34.245.231Source IP Port (Media): 20500Destn IP Address (Media): 166.34.245.230Destn IP Port (Media): 20644Destn SIP Addr (Control) : 166.34.245.230Destn SIP Port (Control) : 5060Destination Name : 166.34.245.230*Mar 8 17:38:38:Disconnect Cause (CC) : 16Disconnect Cause (SIP) : 200Related Commands
Enables all SIP-related debugging.
Shows SIP SPI errors.
Shows all SIP SPI events tracing.
Shows all SIP SPI message tracing.
Shows all SIP SPI state tracing.
debug ccsip error
To show SIP SPI errors, enter the debug ccsip error command in privileged EXEC configuration mode. To disable debugging output, use the no form of this command.
debug ccsip error
no debug ccip error
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command traces all error messages generated from errors encountered by the SIP subsystem.
Examples
From one side of the call, the debug output is as follows:
Router1# debug ccsip errorSIP Call error tracing is enabledRouter1#*Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_idle_call_setup*Mar 6 14:16:41: act_idle_call_setup:Not using Voice Class Codec*Mar 6 14:16:41: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 160*Mar 6 14:16:41: REQUEST CONNECTION TO IP:166.34.245.231 PORT:5060*Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_idle_connection_created*Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 166.34.245.231:5060, local_port 55674*Mar 6 14:16:41: sipSPIAddLocalContact*Mar 6 14:16:41: CCSIP-SPI-CONTROL: sip_stats_method*Mar 6 14:16:41: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060*Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_sentinvite_new_message*Mar 6 14:16:41: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 6 14:16:41: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:16:41: Roundtrip delay 4 milliseconds for method INVITE*Mar 6 14:16:41: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060*Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_recdproc_new_message*Mar 6 14:16:41: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 6 14:16:41: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description*Mar 6 14:16:41: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:16:41: Roundtrip delay 8 milliseconds for method INVITE*Mar 6 14:16:41: HandleSIP1xxRinging: SDP MediaTypes negotiation successful!Negotiated Codec : g711ulaw , bytes :160Inband Alerting : 0*Mar 6 14:16:45: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060*Mar 6 14:16:45: CCSIP-SPI-CONTROL: act_recdproc_new_message*Mar 6 14:16:45: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 6 14:16:45: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description*Mar 6 14:16:45: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:16:45: Roundtrip delay 3844 milliseconds for method INVITE*Mar 6 14:16:45: CCSIP-SPI-CONTROL: act_recdproc_new_message: SDP MediaTypes negotiation successful!Negotiated Codec : g711ulaw , bytes :160*Mar 6 14:16:45: CCSIP-SPI-CONTROL: sipSPIReconnectConnection*Mar 6 14:16:45: CCSIP-SPI-CONTROL: recv_200_OK_for_invite*Mar 6 14:16:45: CCSIP-SPI-CONTROL: sip_stats_method*Mar 6 14:16:45: HandleUdpReconnection: Udp socket connected for fd: 1 with 166.34.245.231:5060*Mar 6 14:16:45: CCSIP-SPI-CONTROL: ccsip_caps_ind*Mar 6 14:16:45: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160*Mar 6 14:16:45: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE*Mar 6 14:16:45: CCSIP-SPI-CONTROL: ccsip_caps_ack*Mar 6 14:16:49: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:56101*Mar 6 14:16:49: CCSIP-SPI-CONTROL: act_active_new_message*Mar 6 14:16:49: CCSIP-SPI-CONTROL: sact_active_new_message_request*Mar 6 14:16:49: CCSIP-SPI-CONTROL: sip_stats_method*Mar 6 14:16:49: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 6 14:16:49: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(16) for outgoing call*Mar 6 14:16:49: CCSIP-SPI-CONTROL: act_disconnecting_disconnect*Mar 6 14:16:49: CCSIP-SPI-CONTROL: sipSPICallCleanup*Mar 6 14:16:49: CLOSE CONNECTION TO CONNID:1*Mar 6 14:16:49: sipSPIIcpifUpdate :CallState: 4 Playout: 2945 DiscTime:48340988 ConnTime 48340525*Mar 6 14:16:49: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060Router1#From the other side of the call, the debug output is as follows:
Router2# debug ccsip errorSIP Call error tracing is enabledRouter2#*Mar 8 17:42:39: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:55674*Mar 8 17:42:39: CCSIP-SPI-CONTROL: sipSPISipIncomingCall*Mar 8 17:42:39: CCSIP-SPI-CONTROL: act_idle_new_message*Mar 8 17:42:39: CCSIP-SPI-CONTROL: sact_idle_new_message_invite*Mar 8 17:42:39: CCSIP-SPI-CONTROL: sip_stats_method*Mar 8 17:42:39: sact_idle_new_message_invite:Not Using Voice Class Codec*Mar 8 17:42:39: sact_idle_new_message_invite: Preferred codec[0] type: g711ulaw Bytes :160*Mar 8 17:42:39: sact_idle_new_message_invite: Media Negotiation successful for anincoming call*Mar 8 17:42:39: sact_idle_new_message_invite: Negotiated Codec : g711ulaw, bytes :160Preferred Codec : g711ulaw, bytes :160*Mar 8 17:42:39: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:42:39: Num of Contact Locations 1 3660110 166.34.245.230 5060*Mar 8 17:42:39: CCSIP-SPI-CONTROL: act_recdinvite_proceeding*Mar 8 17:42:39: CCSIP-SPI-CONTROL: ccsip_caps_ind*Mar 8 17:42:39: ccsip_caps_ind: codec(negotiated) = 5(Bytes 160)*Mar 8 17:42:39: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160*Mar 8 17:42:39: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE*Mar 8 17:42:39: CCSIP-SPI-CONTROL: ccsip_caps_ack*Mar 8 17:42:39: CCSIP-SPI-CONTROL: act_recdinvite_alerting*Mar 8 17:42:39: 180 Ringing with SDP - not likely*Mar 8 17:42:39: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:42:42: CCSIP-SPI-CONTROL: act_sentalert_connect*Mar 8 17:42:42: sipSPIAddLocalContact*Mar 8 17:42:42: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:42:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:55674*Mar 8 17:42:42: CCSIP-SPI-CONTROL: act_sentsucc_new_message*Mar 8 17:42:42: CCSIP-SPI-CONTROL: sip_stats_method*Mar 8 17:42:47: CCSIP-SPI-CONTROL: act_active_disconnect*Mar 8 17:42:47: REQUEST CONNECTION TO IP:166.34.245.230 PORT:5060*Mar 8 17:42:47: CCSIP-SPI-CONTROL: act_active_connection_created*Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection*Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection: Connid(1) created to 166.34.245.230:5060, local_port 56101*Mar 8 17:42:47: CCSIP-SPI-CONTROL: sip_stats_method*Mar 8 17:42:47: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:55674*Mar 8 17:42:47: CCSIP-SPI-CONTROL: act_disconnecting_new_message*Mar 8 17:42:47: CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response*Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICheckResponse*Mar 8 17:42:47: CCSIP-SPI-CONTROL: sip_stats_status_code*Mar 8 17:42:47: Roundtrip delay 0 milliseconds for method BYE*Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICallCleanup*Mar 8 17:42:47: CLOSE CONNECTION TO CONNID:1*Mar 8 17:42:47: sipSPIIcpifUpdate :CallState: 4 Playout: 1255 DiscTime:66856757 ConnTime 66856294*Mar 8 17:42:47: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060Related Commands
debug ccsip events
To show all SIP SPI events tracing, enter the debug ccsip events command in privileged EXEC configuration mode. To disable debugging output, use the no form of this command.
debug ccsip events
no debug ccsip events
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command traces the events posted to SIP SPI from all interfaces.
Examples
From one side of the call, the debug output is as follows:
Router1# debug ccsip eventsSIP Call events tracing is enabledRouter1#*Mar 6 14:17:57: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP*Mar 6 14:17:57: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION*Mar 6 14:17:57: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 6 14:18:00: Queued event from SIP SPI : SIPSPI_EV_RECONNECT_CONNECTION*Mar 6 14:18:00: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 6 14:18:04: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 6 14:18:04: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT*Mar 6 14:18:04: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTIONRouter1#From the other side of the call, the debug output is as follows:
Router2# debug ccsip eventsSIP Call events tracing is enabledRouter2#*Mar 8 17:43:55: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:43:55: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING*Mar 8 17:43:55: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_ALERTING*Mar 8 17:43:55: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:43:58: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_CONNECT*Mar 8 17:43:58: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:44:01: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT*Mar 8 17:44:01: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION*Mar 8 17:44:01: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE*Mar 8 17:44:01: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTIONRelated Commands
debug ccsip messages
To show all SIP SPI message tracing, enter the debug ccsip messages command in privileged EXEC configuration mode. To disable debugging output, use the no form of this command.
debug ccsip messages
no debug ccsip messages
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.
Examples
From one side of the call, the debug output is as follows:
Router1# debug ccsip messageSIP Call messages tracing is enabledRouter1#*Mar 6 14:19:14: Sent:INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Sat, 06 Mar 2002 19:19:14 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Cisco-Guid: 2881152943-2184249568-0-483551624User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEMax-Forwards: 6Timestamp: 731427554Contact: <sip:3660110@166.34.245.230:5060;user=phone>Expires: 180Content-Type: application/sdpContent-Length: 138v=0o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20762 RTP/AVP 0*Mar 6 14:19:14: Received:SIP/2.0 100 TryingVia: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:45:12 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Timestamp: 731427554Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Length: 0*Mar 6 14:19:14: Received:SIP/2.0 180 RingingVia: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:45:12 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Timestamp: 731427554Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Type: application/sdpContent-Length: 138v=0o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20224 RTP/AVP 0*Mar 6 14:19:16: Received:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357Date: Mon, 08 Mar 2002 22:45:12 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Timestamp: 731427554Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledContact: <sip:3660210@166.34.245.231:5060;user=phone>CSeq: 101 INVITEContent-Type: application/sdpContent-Length: 138v=0o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20224 RTP/AVP 0*Mar 6 14:19:16: Sent:ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357Date: Sat, 06 Mar 2002 19:19:14 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Max-Forwards: 6Content-Type: application/sdpContent-Length: 138CSeq: 101 ACKv=0o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20762 RTP/AVP 0*Mar 6 14:19:19: Received:BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.231:53600From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357To: "3660110" <sip:3660110@166.34.245.230>Date: Mon, 08 Mar 2002 22:45:14 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledMax-Forwards: 6Timestamp: 731612717CSeq: 101 BYEContent-Length: 0*Mar 6 14:19:19: Sent:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.231:53600From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357To: "3660110" <sip:3660110@166.34.245.230>Date: Sat, 06 Mar 2002 19:19:19 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledTimestamp: 731612717Content-Length: 0CSeq: 101 BYERouter1#From the other side of the call, the debug output is as follows:
Router2# debug ccsip messageSIP Call messages tracing is enabledRouter2#*Mar 8 17:45:12: Received:INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Sat, 06 Mar 2002 19:19:14 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Cisco-Guid: 2881152943-2184249568-0-483551624User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEMax-Forwards: 6Timestamp: 731427554Contact: <sip:3660110@166.34.245.230:5060;user=phone>Expires: 180Content-Type: application/sdpContent-Length: 138v=0o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20762 RTP/AVP 0*Mar 8 17:45:12: Sent:SIP/2.0 100 TryingVia: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:45:12 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Timestamp: 731427554Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Length: 0*Mar 8 17:45:12: Sent:SIP/2.0 180 RingingVia: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>Date: Mon, 08 Mar 2002 22:45:12 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Timestamp: 731427554Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledCSeq: 101 INVITEContent-Type: application/sdpContent-Length: 138v=0o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20224 RTP/AVP 0*Mar 8 17:45:14: Sent:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357Date: Mon, 08 Mar 2002 22:45:12 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Timestamp: 731427554Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledContact: <sip:3660210@166.34.245.231:5060;user=phone>CSeq: 101 INVITEContent-Type: application/sdpContent-Length: 138v=0o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231s=SIP Callt=0 0c=IN IP4 166.34.245.231m=audio 20224 RTP/AVP 0*Mar 8 17:45:14: Received:ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.230:55820From: "3660110" <sip:3660110@166.34.245.230>To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357Date: Sat, 06 Mar 2002 19:19:14 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Max-Forwards: 6Content-Type: application/sdpContent-Length: 138CSeq: 101 ACKv=0o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230s=SIP Callt=0 0c=IN IP4 166.34.245.230m=audio 20762 RTP/AVP 0*Mar 8 17:45:17: Sent:BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 166.34.245.231:53600From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357To: "3660110" <sip:3660110@166.34.245.230>Date: Mon, 08 Mar 2002 22:45:14 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledMax-Forwards: 6Timestamp: 731612717CSeq: 101 BYEContent-Length: 0*Mar 8 17:45:17: Received:SIP/2.0 200 OKVia: SIP/2.0/UDP 166.34.245.231:53600From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357To: "3660110" <sip:3660110@166.34.245.230>Date: Sat, 06 Mar 2002 19:19:19 GMTCall-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledTimestamp: 731612717Content-Length: 0CSeq: 101 BYERelated Commands
debug ccsip states
To show all SIP SPI state tracing, enter the debug ccsip states command in privileged EXEC configuration mode. To disable debugging output, use the no form of this command.
debug ccsip states
no debug ccsip states
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command traces the state machine changes of SIP SPI and displays the state transitions.
Examples
The following example shows all SIP SPI state tracing:
Router1# debug ccsip statesSIP Call states tracing is enabledRouter1#*Jan 2 18:34:37.793:0x6220C634 :State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)*Jan 2 18:34:37.797:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING)*Jan 2 18:34:37.797:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING)*Jan 2 18:34:37.801:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE)*Jan 2 18:34:37.809:0x6220C634 :State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)*Jan 2 18:34:37.853:0x6220C634 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING)*Jan 2 18:34:38.261:0x6220C634 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) to (STATE_ACTIVE, SUBSTATE_NONE)*Jan 2 18:35:09.860:0x6220C634 :State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)*Jan 2 18:35:09.868:0x6220C634 :State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)*Jan 2 18:28:38.404: Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTIONRelated Commands
default
To reset the value of a SIP-related command to its default, enter the default command in SIP user agent configuration mode. To disable the default setting, use the no form of this command.
default {inband-alerting | max-forwards | retry {invite | response | bye | cancel} | sip-server | timers {trying | connect | disconnect | expires} | | transport}
no default {inband-alerting | max-forwards | retry {invite | response | bye | cancel} | sip-server | timers {trying | connect | disconnect | expires} | | transport}
Syntax Description
Defaults
No default behavior or values.
Command Modes
SIP user agent configuration
Command History
Examples
The following example shows how to set inband-alerting to default value:
Router(config-sip-ua)# default inband-alertingRelated Commands
gw-accounting
To enable gateway-specific accounting, enter the gw-accounting command in global configuration mode. To disable gateway-specific accounting, use the no form of this command.
gw-accounting {h323 [vsa] | syslog | voip}
no gw-accounting {h323 [vsa] | syslog | voip}
Syntax Description
Defaults
Disabled.
Command Modes
Global configuration
Command History
Usage Guidelines
There are three different methods of accounting:
•
•
•
Use this command if you configure the AAA accounting application. If you enable both h323 and syslog simultaneously, CDRs are generated in both methods.
To collect basic start-stop connection accounting data, the gateway must be configured to support gateway-specific H.323 accounting functionality. The gw-accounting command enables you to send accounting data to the RADIUS server in one of four ways:
•
For more information about RADIUS and the use of IETF-defined attributes, refer to the Cisco IOS Security Configuration Guide.
•
<session id>/<call leg setup time>/<gateway id>/<connection id>/<call origin>/ <call type>/<connect time>/<disconnect time>/<disconnect cause>/<remote ip address>Table 5 shows the field attributes that you use with the overloaded session-ID method and a brief description of each.
Because of the limited size of the Acct-Session-Id string, it is not possible to embed very many information elements in it. Therefore, this feature supports only a limited set of accounting information elements.
Use the gw-accounting h323 command to configure the overloaded session ID method of applying H.323 gateway-specific accounting.
•
protocol: attribute sep value *"Protocol" is a value of the Cisco "protocol" attribute for a particular type of authorization. "Attribute" and "value" are an appropriate attribute/value (AV) pair defined in the Cisco TACACS+ specification, and "sep" is "=" for mandatory attributes and "*" for optional attributes. This allows the full set of features available for TACACS+ authorization to also be used for RADIUS.
The VSA fields and their ASCII values are listed in Table 6.
Use the gw-accounting h323 vsa command to configure the VSA method of applying H.323 gateway-specific accounting.
•
<server timestamp> <gateway id> <message number> : <message label> : <list of AV pairs>The syslog message fields are listed in Table 7.
Use the gw-accounting syslog command to configure the syslog record method of gathering H.323 accounting data.
Use this command if you configure the AAA accounting application.
If you enable both h323 and syslog simultaneously, CDRs are generated in both methods.
Examples
The following example shows how to configure accounting using RADIUS to output accounting CDRs. Both H.323 and SIP protocols can use this method.
Router(config)# gw-accounting voipThe following example configures basic H.323 accounting using IETF RADIUS attributes:
gw-accounting h323The following example configures H.323 accounting using VSA RADIUS attributes:
gw-accounting h323 vsaThe following example enables gateway-specific accounting and defines the accounting method as voip:
gw-accounting voipRelated Commands
Enables inband alerting so that the originating gateway can open an early media path (upon receiving a 180 or 183 message with a SDP body).
inband-alerting
To enable inband alerting, enter the inband-alerting command in the SIP user agent configuration mode. Use the no form of this command to disable inband alerting.
[no] inband-alerting
Syntax Description
There are no arguments or keywords for this command.
Defaults
By default, inband alerting is enabled.
Command Modes
SIP user agent configuration
Command History
Usage Guidelines
If inband alerting is enabled, the originating gateway can open an early media path (upon receiving a 180 or 183 message with a SDP body). This allows the terminating gateway or switch to feed tones or announcements before the call is connected. If inband-alerting is disabled, local alerting is generated on the originating gateway.
To reset this command to the default value, use the default command.
Examples
The following example shows how to disable inband alerting:
Router(config)# sip-uaRouter(config-sip-ua)# no inband-alertingRelated Commands
max-forwards
To set the maximum number of proxy or redirect servers that can forward a request, enter the max-forwards command in SIP user agent configuration mode. To reset this command to the default value, use the no form of this command.
max-forwards number
no max-forwards number
Syntax Description
Defaults
The default number of hops is 6.
Command Modes
SIP user agent configuration
Command History
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
The following example shows how to set the number of proxy or redirect servers that can forward a request to two:
Router(config)# sip-uaRouter(config-sip-ua)# max-forwards 2Related Commands
max-redirects
To set the maximum number of redirect servers that a call can traverse, enter the max-redirects command in dial-peer configuration mode. To reset this command to the default value, use the no form of this command.
max-redirects number
no max-redirects number
Syntax Description
number
Maximum number of redirect servers that a call can traverse. Possible values are 1 through 10. The default is 1.
Defaults
The default number of redirects is 1.
Command Modes
Dial-peer configuration
Command History
Examples
The following example shows how to set the number of redirect servers that a call can traverse to one:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# max-redirects 2Related Commands
session protocol
To configure a VoIP dial peer to use either H323 or SIP as the session protocol for VoIP call signaling, enter the session protocol command in dial-peer configuration mode. To reset to the default, use the no form of this command.
session protocol {aal2-trunk | cisco | sipv2 | smtp}
no session protocol
Syntax Description
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
The cisco keyword is applicable only to VoIP on the Cisco 1750, Cisco 1751, Cisco 3600 series, and Cisco 7200 series routers.
The aal2-trunk keyword is applicable only to VoATM on the Cisco MC3810 multiservice access concentrator and the Cisco 7200 series router.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example shows how to configure the dial peer to use IETF SIP:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# session protocol sipv2The following example shows that AAL2 trunking has been configured as the session protocol:
dial-peer voice 10 voatmsession protocol aal2-trunkThe following example shows that Cisco session protocol has been configured as the session protocol:
dial-peer voice 20 voipsession protocol ciscoRelated Commands
Specifies a network-specific address for a dial peer.
Configures the VoIP dial peer to use TCP or UDP as the underlying transport layer protocol for SIP messages.
session target (VoIP)
To designate a network-specific address to receive calls from this VoIP dial peer, use the session target command in dial-peer configuration mode. To reset to the default, use the no form of this command.
Cisco 1751, Cisco 3725, Cisco 3745, Cisco AS5300
session target {ipv4:destination-address | dns:[$s$. | $d$. | $u$. | $e$.] host-name | enum:table-num | loopback:rtp | ras | sip-server}
no session target
Cisco 2600 Series, Cisco 3600 Series, Cisco AS5350, Cisco AS5400, Cisco AS5850, Cisco MC8310
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | enum:table-num | loopback:rtp | ras | settlement provider-number | sip-server}
no session target
Cisco AS5800
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | enum:table-num | loopback:rtp }
no session target
Syntax Description
Defaults
Enabled, with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use this command to specify a network-specific destination for a dial peer to receive calls from this dial peer. You can select an option to define a network-specific address or domain name as a target, or you can select one of several methods to automatically determine the destination for calls from this dial peer.
Use the session target dns command with or without the specified macros. Using the optional macros can reduce the number of VoIP dial peer session targets you must configure if you have groups of numbers associated with a particular router.
The session target enum command instructs the dial peer to use a table of translation rules to convert the dialed number identification service (DNIS) number into a number in E.164 format. This translated number is sent to a DNS server that contains a collection of URLs. These URLs identify each user as a destination for a call and may represent various access services, such as SIP, H.323, telephone, fax, email, instant messaging, and personal web pages. Before assigning the session target to the dial peer, configure an ENUM match table with the translation rules using the voice enum-match-table command in global configuration mode. The table is identified in session target enum as table-num.
Use the session target loopback command to test the voice transmission path of a call. The loopback point depends on the call origin.
Use the session target ras command to specify that the RAS protocol is being used to determine the IP address of the session target.
In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of RAS with the settle-call command.
If the session target type is settlement when the VoIP dial peers are configured for a settlement server, the provider-number parameter in the session target and settle-call commands should be identical.
Use the session target sip-server command to name the global SIP server interface as the destination for calls from this dial peer. You must first define the SIP server interface by using the sip-server command in SIP user-agent configuration mode. Then you can enter the session target sip-server option for each dial peer instead of having to enter the entire IP address for the SIP server interface under each dial peer.
Examples
The following example creates a session target using DNS for a host named "voice_router" in the domain cisco.com:
dial-peer voice 10 voipsession target dns:voice_router.cisco.comThe following example creates a session target using DNS with the optional $u$. macro. In this example, the destination pattern ends with four periods (.) to allow for any four-digit extension that has the leading numbers 1310222.
The optional macro $u$. directs the gateway to use the unmatched portion of the dialed number—in this case, the four-digit extension—to identify a dial peer. As in the preceding example, the domain is "cisco.com."
dial-peer voice 10 voipdestination-pattern 1310222....session target dns:$u$.cisco.comThe following example creates a session target using DNS, with the optional $d$. macro. In this example, the destination pattern has been configured for 13102221111. The optional macro $d$. directs the gateway to use the destination pattern to identify a dial peer in the "cisco.com" domain.
dial-peer voice 10 voipdestination-pattern 13102221111session target dns:$d$.cisco.comThe following example creates a session target using DNS, with the optional $e$. macro. In this example, the destination pattern has been configured for 12345. The optional macro $e$. directs the gateway to do the following: reverse the digits in the destination pattern, add periods between the digits, and use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voipdestination-pattern 12345session target dns:$e$.cisco.comThe following example creates a session target using an ENUM table. It indicates that calls made using dial peer 100 should use the preferential order of rules in enum match table 3.
dial-peer voice 101 voipsession target enum: 3The following example creates a session target using RAS:
dial-peer voice 11 voipdestination-pattern 13102221111session target rasThe following example creates a session target using settlement:
dial-peer voice 24 voipsession target settlement:0Related Commands
session transport
To configure the VoIP dial peer to use TCP or UDP as the underlying transport layer protocol for SIP messages, enter the session transport command in dial-peer configuration mode. To reset the value of this command to the default, use the no form of this command.
session transport {udp | tcp}
no session transport {udp | tcp}
Syntax Description
udp
Configures the SIP dial peer to use the UDP transport layer protocol. This is the default.
tcp
Configures the SIP dial peer to use the TCP transport layer protocol.
Defaults
The default for this command is that the SIP dial peer uses UDP.
Note
session transport command must be the same.Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use the show sip-ua status command in privileged EXEC configuration mode to ensure that the transport protocol that you set using the session transport command matches the protocol set using the transport command. This command is used in a dial-peer configuration mode to specify the SIP transport method, either UDP or TCP.
Examples
The following example shows how to configure the SIP dial peer to use the UDP transport layer protocol:
Router(config)# dial-peer voice 102 voipRouter(dial-peer-config)# session transport udpRelated Commands
Configures a VoIP dial peer to use either H323 or SIP as the session protocol for VoIP call signaling
Specifies a network-specific address for a dial peer.
show sip-ua statistics
To display response, traffic, and retry SIP UA statistics, enter the show sip-ua statistics command in privileged EXEC configuration mode.
show sip-ua statistics
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use this command to verify SIP configurations.
Examples
The following example shows response, traffic, and retry SIP UA statistics:
Router# show sip-ua statisticsSIP Response Statistics (Inbound/Outbound)Informational:Trying 0/0, Ringing 0/0,Forwarded 0/0, Queued 0/0,SessionProgress 0/0Success:OkInvite 0/0, OkBye 0/0,OkCancel 0/0, OkOptions 0/0Redirection (Inbound only):MultipleChoice 0, MovedPermanently 0,MovedTemporarily 0, SeeOther 0,UseProxy 0, AlternateService 0Client Error:BadRequest 0/0, Unauthorized 0/0,PaymentRequired 0/0, Forbidden 0/0,NotFound 0/0, MethodNotAllowed 0/0,NotAcceptable 0/0, ProxyAuthReqd 0/0,ReqTimeout 0/0, Conflict 0/0, Gone 0/0,LengthRequired 0/0, ReqEntityTooLarge 0/0,ReqURITooLarge 0/0, UnsupportedMediaType 0/0,BadExtension 0/0, TempNotAvailable 0/0,CallLegNonExistent 0/0, LoopDetected 0/0,TooManyHops 0/0, AddrIncomplete 0/0,Ambiguous 0/0, BusyHere 0/0Server Error:InternalError 0/0, NotImplemented 0/0,BadGateway 0/0, ServiceUnavail 0/0,GatewayTimeout 0/0, BadSipVer 0/0Global Failure:BusyEverywhere 0/0, Decline 0/0,NoExistAnywhere 0/0, NotAcceptable 0/0SIP Total Traffic Statistics (Inbound/Outbound)Invite 0/0, Ack 0/0, Bye 0/0,Cancel 0/0, Options 0/0Retry StatisticsInvite 0, Bye 0, Cancel 0, Response 0
Related Commands
show sip-ua status
To display SIP UA status, enter the show sip-ua status command in privileged EXEC configuration mode to display SIP status.
show sip-ua status
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use this command to verify SIP configurations.
Examples
The following example displays SIP UA status:
Router# show sip-ua statusSIP User Agent StatusSIP User Agent for UDP : ENABLEDSIP User Agent for TCP : ENABLEDSIP max-forwards :6Table 9 describes significant fields in this output.
Related Commands
Displays response, traffic, and retry SIP UA statistics.
Displays the current settings for SIP UA timers.
show sip-ua timers
To display the current settings for SIP UA timers. enter the show sip-ua timers command in privileged EXEC configuration mode.
show sip-ua timers
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Examples
The following examples displays SIP UA timers values:
Router# show sip-ua timersSIP UA Timer Values (millisecs)trying 500, expires 150000, connect 500, disconnect 500comet 500, prack 500, rel1xx 500, notify 500Table 10 describes significant fields in this output.
Related Commands
sip-server
To configure a network address for the SIP server interface, enter the sip-server command in SIP user agent configuration mode. To disable, use the no form of this command.
sip-server {dns:[host-name] | ipv4:ipaddr [:port-num]}
no sip-server {dns:[host-name] | ipv4:ipaddr [:port-num]}
Syntax Description
Defaults
The default for this command is a null value.
Command Modes
SIP user agent configuration
Command History
Usage Guidelines
You can specify session target sip-server for each dial peer instead of repeatedly entering the SIP server interface address for each dial peer. To reset this command to a null value, use the default command.
Examples
The following example shows how to set the global SIP server interface to a DNS host name and specify an IP address. If you do not specify a host name, the default DNS defined by the ip name-server command is used.
Router(config)# sip-uaRouter(config-sip-ua)# sip-server dns:UA-1-f0.sip.comRelated Commands
sip-ua
To enable the sip-ua configuration commands to configure the user agent, enter the sip-ua command in global configuration mode. To reset all configuration commands to their default values, use the no form of this command.
sip-ua
no sip-ua
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Usage Guidelines
Use the sip-ua command to enter the SIP user agent-configuration sub-mode. The table below lists the sub-mode configuration commands.
Examples
The following example shows sub-command options available in config-sip-ua configuration mode:
Router(config)# sip-uaRouter(config-sip-ua)# ?SIP UA configuration commands:default Set a command to its defaultsexit Exit from sip-ua configuration modeinband-alerting Specify an Inband-alerting SIP headermax-forwards Change number of max-forwards for SIP Methodsno Negate a command or set its defaultsretry Change default retries for each SIP Methodsip-server Configure a SIP Server Interfacetimers SIP Signaling Timers Configurationtransport Enable SIP UA transport for TCP/UDPRelated Commands
timers
To configure the SIP signaling timers, enter the timers command in SIP user agent configuration mode. To reset to the default value, use the no form of this command.
timers {trying number | connect number | disconnect number | expires number}
no timers
Syntax Description
Defaults
The default for trying, connect, and disconnect is 500. The default for expires is 180,000.
Command Modes
SIP user agent configuration
Command History
Usage Guidelines
If you used an earlier version of this command to configure timers, the timer settings are maintained. The output of the show running configuration command reflects both previous and current timers.
To reset this command to the default value, you can also use the default command.
Examples
The following example sets the trying timers to the default of 500.
Router(config)# sip-uaRouter(config-sip-ua)# timers trying 500Related Commands
transport
To configure the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket, enter the transport command in SIP user agent configuration mode. To block reception of SIP signaling messages on a particular socket, use the no form of this command.
transport {udp | tcp}
no transport {udp | tcp}
Syntax Description
udp
Configures the SIP user agent to receive SIP messages on UDP port 5060.
tcp
Configures the SIP user agent to receive SIP messages on TCP port 5060.
Defaults
By default, both UDP and TCP transport protocols are enabled.
Command Modes
SIP user agent configuration
Command History
Usage Guidelines
This command controls whether messages reach the SIP service provider interface (SPI). To reset this command to the default value, use the default command.
Examples
The following example shows how to block reception of SIP signaling messages on a TCP port:
Router(config)# sip-uaRouter(config-sip-ua)# no transport tcpRelated Commands
Enables the sip-ua configuration commands to configure the user agent.
Configures a network address for the SIP server interface.
Glossary
AAA—Authentication, Authorization, and Accounting. AAA is a suite of network security services that provides the primary framework through which access control can be set up on your Cisco router or access server.
ANI—Automatic number identification.
call—In SIP, a call consists of all participants in a conference invited by a common source. A SIP call is identified by a globally unique call ID. A point-to-point IP telephony conversation maps into a single SIP call. For a multicast session, each participant in the session constitutes a unique call. Each call involves a UAC and a UAS application.
CAS—Channel associated signaling.
CCAPI—Call control applications programming interface.
CLI—Command line interface.
CO—Central office.
CPE—Customer premises equipment. Terminating equipment, such as terminals, telephones, and modems, supplied by the telephone company, installed at the customer sites, and connected to the telephone company network.
CSM—Call switching module.
dial peer—An addressable call endpoint. In Voice over IP (V0IP), there are two types of dial peers: POTS and VoIP.
DNS—Domain name system used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in the locating remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.
DNIS—Dialed number identification service (the called number).
DSP—Digital signal processor.
DTMF—Dual tone multi-frequency.
E.164—The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.
E&M—Ear and mouth RBS signaling.
endpoint—A H.323 terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream.
gateway—A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols by converting protocols. A gateway is the point where a circuit-switched call is encoded and repackaged into IP packets.
H.323—An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.
H.323 RAS—Registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper.
IPSEC—An IETF standard that is used to provide security for transmission of sensitive information over unprotected networks such as the Internet. IPSec acts at the network layer, protecting and authenticating IP packets between participating IPSec devices ("peers"), such as Cisco routers.
IVR—Integrated voice response. When someone dials in, IVR responds with a prompt to get a personal identification number (PIN), and so on.
LEC—Local exchange carrier.
Location Server—A SIP redirect or proxy server uses a a location service to get information about a caller's location(s). Location services are offered by location servers.
MF—Multi-frequency tones are made of six frequencies that provide 15 two frequency combinations for indication digits 0-9 and KP/ST signals.
multicast—A process of transmitting PDUs from one source to many destinations. The actual mechanism (that is, IP multicast, multi-unicast, and so forth) for this process might be different for LAN technologies.
multipoint-unicast—A process of transferring PDUs (Protocol Data Units) where an endpoint sends more than one copy of a media stream to different endpoints. This can be necessary in networks which do not support multicast.
node—A H.323 entity that uses RAS to communicate with the gatekeeper, for example, an endpoint such as a terminal, proxy, or gateway.
PDU—Protocol data units used by bridges to transfer connectivity information.
POTS—Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN.
Proxy Server—An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.
PSTN—Public switched telephone network. PSTN refers to the local telephone company.
QoS—Quality of Service. Measure of performance for a transmission system that reflects its transmission quality and service availability. QoS refers to the ability of a network to provide better service to selected network traffic over various underlying technologies. QoS is not inherent in a network infrastructure. Rather, you must institute QoS by strategically deploying features that implement it throughout the network.
Redirect Server—A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client. It does not initiate its own SIP request nor accept calls.
Registrar—A registrar is a server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and MAY offer location services.
RAS—Registration, admission, and status protocol. This is the protocol that is used between endpoints and the gatekeeper to perform management functions.
RBS—Robbed bit signaling.
session—In SIP, a session is a set of multimedia senders and receivers and the data streams flowing between the senders and receivers. A SIP multimedia conference is an example of a session. A caller can be invited several times, by different calls, to the same session.
SIP—Session Initiation Protocol. This is an application-layer protocol developed by the IETF MMUSIC Working Group to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP features are compliant with IETF RFC 2543, published in March 1999.
SPI—Service provider interface.
TDM—Time division multiplexing. Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.
User Agent—see UAS and UAC.
UAC—User Agent Client: A user agent client is a client application that initiates the SIP request.
UAS—User Agent Server (or user agent): A user agent server is a server application that contacts the user when a SIP request is received, then returns a response on behalf of the user. The response accepts, rejects or redirects the request.
VoIP—Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTs-like functionality, reliability, and voice quality. VoIP is a blanket term, which generally refers to Cisco's standards based (for example H.323) approach to IP voice traffic.
Guest
