Session Initiation Protocol for VoIP on Cisco Access Platforms [Cisco IOS Software Releases 12.1 T]

Table Of Contents

Session Initiation Protocol for Voice over IP on Cisco Access Platforms

Feature Overview

Benefits

Restrictions

Related Features and Technologies

DNS SRV

Related Documents

Supported Platforms

Supported Standards, MIBs, and RFCs

Standards

MIBs

RFCs

Prerequisites

Configuration Tasks

Configuring the SIP User Agent (UA)

Configuring a VoIP Dial Peer

Configuring a POTS Dial Peer

Verifying the SIP Feature Configuration

Configuration Examples

Basic SIP Configuration

Configuring SIP with Multiple Codecs

Command Reference

default

exit sip-ua

gw-accounting

inband-alerting

max-redirects

retry

session protocol

session target

session transport

show sip-ua

sip-server

sip-ua

timers

transport

Debug Commands

debug ccsip all

debug ccsip calls

debug ccsip error

debug ccsip events

debug ccsip messages

debug ccsip states

Glossary

Session Initiation Protocol for Voice over IP on Cisco Access Platforms

Document Update Alert

This document was originally produced for Cisco IOS Release 12.2(2)XB. This feature has been updated in subsequent releases, and more recent documentation is available.

If you are using Cisco IOS Release 12.3 or higher, refer to the following documentation in the Cisco IOS Voice Configuration Library, Release 12.3:

• Cisco IOS SIP Configuration Guide

If you are using Cisco IOS Release 12.2 or higher, refer to the following chapter in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2:

• Configuring Session Initiation Protocol for Voice over IP

Feature History

Release

Modification

12.1(1)T

This feature was first introduced.

12.2(2)XA

This feature was integrated in this release to support the AS5400 and AS5350 universal port platforms.

12.2(2)XB1

This feature was implemented on the Cisco AS5850 platform.

Voice over Internet Protocol (VoIP) currently implements ITU's H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. Session Initiation Protocol (SIP) is a new protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999.

The Cisco SIP functionality equips the Cisco AS5300 access server, and the Cisco 2600 and Cisco 3600 series routers to signal the setup of voice and multimedia calls over IP networks; therefore, the SIP feature, introduced in Cisco IOS Release 12.1(1)T, provides an alternative to H.323 within the VoIP internetworking software. The SIP feature also provides nonproprietary advantages in the areas of:

•Protocol extensibility

•System scalability

•Personal mobility services

•Interoperability with different vendors

This document includes the following sections:

• Feature Overview

• Benefits

• Restrictions

• Related Features and Technologies

• Related Documents

• Supported Platforms

• Prerequisites

• Configuration Tasks

• Configuration Examples

• Command Reference

• Debug Commands

• Glossary

Feature Overview

The SIP feature includes the following functionality:

•Configurable in-band alerting

•Ability to specify the maximum number of SIP redirects

•Ability to specify SIP or H.323 on a dial-peer basis

•Support for both UDP and TCP transport layers for SIP messages

•Powerful debugging support

•Support for Domain Name System (DNS) SRV records for resolving SIP server hostnames

•Configurable SIP message timers and retries

Benefits

The SIP feature meets the needs of service providers that use SIP on the gateways of their VoIP network to:

•Enable Cisco voice-enabled platforms to provide RFC 2543 compliant user-agent client gateways.

•Support codecs capable of Carrier-class voice quality

Although SIP is simpler than H.323, SIP provides similar capabilities in:

•System scalability

•End-to-end solutions

•High-density voice gateways

Restrictions

Ensure that your access platform has 16 MB Flash and 64 MB DRAM memory minimum, and that I/O memory is set to either 8 or 16 MB.

Related Features and Technologies

The SIP feature is dependent on the interoperability of Service Provider Features for VoIP.

DNS SRV

Currently you must know the exact address of a server to contact it. SRV records enable administrators to use several servers to provide the same service within a single domain. SRV Resource Records (RRs) allow administrators to define primary and backup servers and move services from host-to-host without affecting service.

SIP Implementation on the gateway follows the methods outlined in Appendix D of RFC 2543 and RFC 2052. The retrieved SRV RRs are sorted based on the Priority field. Then, starting from the Highest Priority level, a server is chosen randomly based on the Weightage assigned to it. The target address for the selected server is resolved using DNS A records. If the selected server fails to provide the service, a new server is chosen within the same Priority level, or the next lower Priority level is chosen if higher priority levels have been exhausted. This is repeated until all the priority levels have been exhausted.

Related Documents

• Session Initiation Protocol Call Flows

• Configuring the Cisco AS5300 for Voice Service Provider Features

•Configuring Voice over IP for the Cisco 2600 and Cisco 3600 Series Routers

•Configuring H.323 VoIP Gateway for Cisco Access Platforms

•Configuring H.323 VoIP Gatekeeper for Cisco Access Platforms

•Service Provider Features for Voice over IP was introduced in Cisco IOS Release 12.0(3)T.

•Voice over IP for the Cisco AS5300 Documents

•Cisco IOS IP and IP Routing Configuration Guide

•Cisco IOS Release Multiservice Applications Configuration Guide

Supported Platforms

•Cisco AS5300

•Cisco 2600 Series

•Cisco 3600 Series

•Cisco AS5400 universal gateway

•Cisco AS5350 universal gateway

•Cisco AS5850 universal gateway

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.

For descriptions of supported MIBs and how to use MIBs, see Cisco's MIB web site on CCO at: http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.

RFCs

RFC 2543 is supported by this feature.

Prerequisites

•Your gateway must have voice functionality that is configurable for either SIP or H.323.

•Establish a working IP network.

For more information about configuring IP, refer to Cisco IOS IP and IP Routing Configuration Guide.

•Configure VoIP.

For more information about configuring VoIP for the appropriate access platform, refer to the Cisco IOS Release Multiservice Applications Configuration Guide.

Configuration Tasks

To configure SIP functions on the Cisco AS5300, or the Cisco 2600 or Cisco 3600 series router, perform the following tasks:

• Configuring the SIP User Agent (UA)

• Configuring a VoIP Dial Peer

• Configuring a POTS Dial Peer

For more information on SIP configuration, including call flows, refer to the Session Initiation Protocol Call Flows document on CCO.

Configuring the SIP User Agent (UA)

A terminating gateway that is not configured as an SIP user agent cannot receive incoming SIP calls. The transport command opens the SIP listener port (5060) to receive SIP (a SIP user agent is configured to listen by default).

To configure the terminating gateway, enter the following commands beginning in global configuration mode:
Command

Purpose
Step 1
Router (config) # sip-ua
Enters the 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.
Step 3
Router (config-sip-ua)# sip-server ipv4:ip-address
Enters the IP address of the SIP server interface.
Step 4
Router (config-sip-ua)# timers invite-wait-100 number 
Sets time to wait for a response.
Step 5
Router (config-sip-ua)# retry invite number
Configures the SIP signaling timers for retry attempts.
Step 6
Router (config-sip-ua)# inband-alerting header-string
Specifies an inband-alerting SIP header.
Configuring a VoIP Dial Peer

When you configure a VoIP dial peer, you must add the following commands:
Command

Purpose
Step 1
Router (config-dial-peer-voice) # dial-peer voice number voip
Enters the dial-peer mode to configure a VoIP dial peer.
Step 2
Router (config-dial-peer-voice) # destination-pattern [+]string[t]
Defines the telephone number associated with this VoIP dial peer.
Step 3
Router (config-dial-peer-voice) # session transport {udp|tcp}
Enters the session transport type for the SIP user agent.
Step 4
Router (config-dial-peer-voice) # session protocol sipv2 
Enters the session protocol type.
Step 5
Router (config-dial-peer-voice) # session target sip-server
Specifies the dial peer session target to use the global SIP server.
Configuring a POTS Dial Peer

When you configure a POTS dial peer, you must add the following commands:
Command

Purpose
Step 1
Router (config-dial-peer-voice) # dial-peer voice number pots
Enters the dial-peer mode to configure a VoIP dial peer.
Step 2
Router (config-dial-peer-voice) # destination-pattern [+]string[t]
Defines the telephone number associated with this POTS dial peer.
Step 3
Router (config-dial-peer-voice) # port 
slot-number/subunit-number/port
Associates this POTS dial peer with a specific voice port.
Step 4
Router (config-dial-peer-voice) # session transport {udp|tcp}
Enters the session transport type for the SIP user agent.
Step 5
Router (config-dial-peer-voice) # session protocol sipv2 
Enters the session protocol type.
Step 6
Router (config-dial-peer-voice) # session target sip-server
Specifies the dial peer session target to use the global SIP server.
Verifying the SIP Feature Configuration

Enter the show running configuration command to verify your configuration.

Configuration Examples

Note All IP addresses and patterns are examples only.

See samples of screen output displays for running configurations:

• Basic SIP Configuration

• Configuring SIP with Multiple Codecs

Basic SIP Configuration

The following shows an example of the output that appears when you enter the show run command. Irrelevant modules are omitted.
version 12.0
. 
. 
.
hostname UA-1
. 
. 
.
ip name-server ip-address
. 
. 
.
isdn switch-type primary-5ess
. 
. 
.
!  The "description" used on the T1 controller will appear in the FROM header
!  of the SIP Invite.
. 
. 
.
controller T1 2
 framing esf
 clock source line secondary 1
 linecode b8zs
 pri-group timeslots 1-24
 description SIP Gateway UA-1; t1-pri controller 2
. 
. 
.
controller T1 3
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
 description SIP Gateway UA-1; t1-pri controller 3
. 
. 
.
voice-port 2:D
. 
. 
.
voice-port 3:D
. 
. 
.
!  Below are examples using the different SIP targets (dns, ipv4, sip-server):
!  on the VOIP dial-peers:
. 
. 
.
dial-peer voice 100 pots
 destination-pattern 9003
 port 3:D
 prefix 9003
. 
. 
.
dial-peer voice 101 voip
 destination-pattern 9004
 session protocol sipv2
 session target sip-server
. 
. 
.
dial-peer voice 200 pots
 destination-pattern 97055500..
 direct-inward-dial
 port 3:D
 prefix 97055500
. 
. 
.
dial-peer voice 201 voip
 destination-pattern 98055500..
 max-redirects 2
 session protocol sipv2
 session target sip-server
 codec g711ulaw
 ip precedence 5
 no vad
. 
. 
.
dial-peer voice 300 pots
 destination-pattern 95055500..
 direct-inward-dial
 port 2:D
 prefix 95055500
. 
. 
.
dial-peer voice 301 voip
 destination-pattern 96055500..
 max-redirects 10
 session protocol sipv2
 session target ipv4:172.16.1.1
 codec g711ulaw
 ip precedence 5
 no vad
!
!  SIP User Agent configuration
!
sip-ua
 retry invite 2
 retry response 2
 retry bye 2
 retry cancel 2
 sip-server ipv4:172.16.1.2
. 
. 
.
interface Ethernet0
 ip address 172.16.1.3 255.255.255.0
 no ip directed-broadcast
 load-interval 30
 no keepalive
 no cdp enable
. 
. 
.
interface Serial2:23
 no ip address
 no ip directed-broadcast
 isdn switch-type primary-5ess
 isdn protocol-emulate user
 isdn incoming-voice modem
 isdn T203 10000
 fair-queue 64 256 0
. 
. 
.
interface Serial3:23
 no ip address
 no ip directed-broadcasT
 isdn switch-type primary-5ess
 isdn protocol-emulate user
 isdn incoming-voice modem
 isdn T203 10000
 fair-queue 64 256 0
. 
. 
.
interface FastEthernet0
 ip address 172.16.1.4 255.255.255.4
 no ip directed-broadcast
 load-interval 30
 duplex auto
 speed auto
. 
. 
.
dialer-list 1 protocol ip permit
Configuring SIP with Multiple Codecs

The following shows an example of the output that appears when you enter the show run command. Inapplicable modules are omitted.
version 12.0
. 
. 
.
hostname UA-4
. 
. 
.
controller T1 0
 framing esf
 clock source line primary
 linecode b8zs
 ds0-group 0 timeslots 1-24 type e&m-fgb dtmf dnis
. 
. 
.
controller T1 1
 framing esf
 clock source line secondary 1
 linecode b8zs
 ds0-group 0 timeslots 1-24 type e&m-fgb dtmf dnis
. 
. 
.
voice-port 0:0
. 
. 
.
voice-port 1:0
. 
. 
.
voice class codec 100
 codec preference 1 g726r16
 codec preference 2 g729r8
 codec preference 3 g711alaw
 codec preference 4 g711ulaw
. 
. 
.
dial-peer voice 500 pots
 destination-pattern 92055500..
 port 0:0
 prefix 92055500
. 
. 
.
dial-peer voice 600 voip
 incoming called-number 92055500..
 session protocol sipv2
 voice-class codec 100
 no vad
. 
. 
.
dial-peer voice 501 pots
 destination-pattern 94055500..
 port 1:0
 prefix 94055500
. 
. 
.
dial-peer voice 601 voip
 incoming called-number 94055500..
 session protocol sipv2
 voice-class codec 100
 no vad
. 
. 
.
interface Ethernet0
 ip address 172.16.1.1 255.255.255.1
 no ip directed-broadcast
 load-interval 30
. 
. 
.
interface FastEthernet0
 ip address 172.16.1.2 255.255.255.2
 no ip directed-broadcast
 load-interval 30
 duplex auto
 speed auto
Command Reference

This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release12.1(1)T command reference publications.

For more information on the search and filter functionality, refer to the Cisco IOS Release 12.1(1)T feature module titled CLI String Search.

This section documents both new and modified commands.

• default

• exit sip-ua

• gw-accounting

• inband-alerting

• max-redirects

• retry

• session protocol

• session target

• session transport

• show sip-ua

• sip-server

• sip-ua

• timers

• transport

default

To reset the value of a command to its default, enter the default SIP user-agent configuration command.

default { inband-alerting | max-forwards | retry {invite | response | bye | cancel } | sip-server | timers { trying | connect | disconnect | expires } | transport }
Syntax Description

inband-alerting

Resets inband-alerting to its default of generating the header "Require: com.cisco.inband-alerting" in outgoing INVITE messages.

max-forwards

Resets max-forwards to its default of 6.

retry {invite | response | bye | cancel }

Resets the specified retry to its default (6 for invite and response; 10 for bye and cancel).

sip-server

Resets the sip-server to a null value.

timers { trying | connect | disconnect | expires }

Resets the specified retry to its default (500 for trying, connect, and disconnect; 180000 for expires).

transport

Resets transport to the default of both UDP and TCP enabled.

Defaults

There are no default behaviors or values for this command.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# sip-ua
Router (config-sip-ua)# default inband-alerting
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

exit sip-ua

To exit the SIP user agent configuration, use the exit sip-ua command in SIP user-agent configuration mode.

exit sip-ua
Syntax Description

This command has no arguments or keywords.

Defaults

There are no default behaviors or values for this command.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# sip-ua
Router (config-sip-ua)# exit sip-ua
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

gw-accounting

To enable gateway specific accounting, use the gw-accounting global configuration command. To disable this function, use the no form of this command.

gw-accounting {voip | syslog | h323 [syslog] }
no gw-accounting {voip | syslog | h323 [syslog] }
Syntax Description

voip

This new option uses RADIUS to output accounting call data records (CDRs). Both H.323 and SIP protocols can use this method, so the name is not bound to a protocol. Use this method with the SIP feature.

syslog

Syslog uses the system logging facility to output CDRs.

h323

H.323 method uses RADIUS to output accounting CDRs.

Defaults

There are no default behaviors or values for this command.

Command Modes

Global configuration

Command History

Release

Modification

11.3(6)NA2

This command was introduced.

12.1(1)T

The voip option was added.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Usage Guidelines

•Use this command if you configure the AAA accounting application.

•Use this command to define a method for accounting and enable accounting on the gateway. There are three accounting methods defined.

•If you enable both h323 and syslog simultaneously, CDRs are generated in both methods.

Examples
Router (config)# gw-accounting voip
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation.

inband-alerting

To specify an inband-alerting SIP header, use the inband-alerting command in SIP user-agent configuration mode. To disable this function, use the no form of this command.

inband-alerting header-string
no inband-alerting header-string
Syntax Description

header-string

Header-string sent to SIP clients to inform them of gateway service provider (SP) behavior.

Defaults

The default generates the header "Require: com.cisco.inband-alerting" in outgoing INVITE messages.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Usage Guidelines

To reset this command to the default value, use the default command.

For more configuration information on inband-alerting, refer to the Session Initiation Protocol Call Flows document.

Examples
Router (config)# sip-ua
Router (config-sip-ua)# inband-alerting 'Cisco inband-alerting required'
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

max-redirects

To set the maximum number of redirects that the user agent allows, use the max-redirects command in the 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
Syntax Description

number

Number of redirects: 1 through 10 are valid inputs.

Defaults

The default number of redirects is 1.

Command Modes

Dial-peer configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# dial-peer voice 102 voip
Router (dial-peer-config)# max-redirects 2
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation.

retry

To configure the retry attempts for SIP messages, use the retry command in SIP user-agent configuration mode. To reset this command to the default value, use the no form of this command.

retry {invite number | response number | bye number | cancel number}
Syntax Description

invite number

Number of INVITE retries: 1 through 10 are valid inputs; default = 6.

response number

Number of RESPONSE retries: 1 through 10 are valid inputs; default = 6.

bye number

Number of BYE retries: 1 through 10 are valid inputs; default = 10.

cancel number

Number of CANCEL retries: 1 through 10 are valid inputs; default = 10.

Defaults

Refer to the Syntax Description table for default values.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# sip-ua
Router (config-sip-ua)# retry invite 5
Related Commands

Command

Description

sip-ua

Enables the sip-ua configuration commands, with which you configure the user agent.

session protocol

To configure a VoIP dial-peer to use either H323 or SIP as the session protocol for VoIP call signaling, use the session protocol command in dial-peer configuration mode. To disable this function, use the no form of this command.

session protocol {cisco | sipv2}
no session protocol {cisco | sipv2}
Syntax Description

cisco

Configure the dial peer to use proprietary Cisco VoIP session protocol.

sipv2

SIP users should use this new option. This option configures the dial peer to use IETF SIP.

Defaults

No default behaviors or values.

Command Modes

Dial-peer configuration

Command History

Release

Modification

11.3(1)T

This command was introduced.

12.0(3)XG

The cisco option was added.

12.1(1)T

The sipv2 option was added.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# dial-peer voice 102 voip
Router (dial-peer-config)# session protocol sipv2 
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation.

session target

To specify a network-specific address for a dial peer, use the session target command in dial-peer configuration mode. To reset this command to the default value, use the no form of this command.

session target { sip-server | dns:host-name | ipv4:ip-address[:port-number] | ras}
no session target
Syntax Description

sip-server

This new option sets the session target to the global SIP server.

dns:host-name

Indicates that the domain name server resolves the name of the IP address. A valid DNS host name is in this form:
gateway.company.com

(Optional) You can use one of the following four wildcards with this keyword when defining the session target for VoIP peers:

•$s$.—Indicates that the source destination pattern is part of the domain name.

•$d$.—Indicates that the destination number is part of the domain name.

•$e$.—Indicates that the digits in the called number are reversed, periods are added in between each digit of the called number, and that this string is part of the domain name.

•$u$.—Indicates that the unmatched portion of the destination pattern (such as a defined extension number) is part of the domain name.

ipv4:ip-address

Sets the IP address of the dial peer. A valid IP address is in this form:
xxx.xxx.xxx.xxx

port-number

(Optional) Contact this port number to complete the call leg.

ras

Enables the Registration, Admission, and Status (RAS) signaling function protocol so that a gatekeeper is consulted to translate the E.164 address to an IP address.

Defaults

The default for this command is enabled with no IP address or domain name defined.

Command Modes

Dial-peer configuration

Command History

Release

Modification

11.3(1)T

This command was introduced.

12.1(1)T

The sip-server option was added.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Usage Guidelines

Enter the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.

You can enter the session target dns command with or without the specified wild cards. Using the optional wildcards can reduce the number of VoIP dial-peer session targets you need to configure if you have groups of numbers associated with a particular router.

Examples
Router (config)# dial-peer voice 102 voip
Router (dial-peer-config)# session target dns:UA-1-f0.sip.com
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation.

sip-server

Configures the SIP server interface.

session transport

To configure the VoIP dial peer to use TCP or UDP as the underlying transport layer protocol for SIP messages, use the session transport command in dial-peer configuration mode. To reset this command to the default value, use the no form of this command.

session transport {udp | tcp}
no session transport
Syntax Description

udp

Configure the SIP dial peer to use the UDP transport layer protocol.

tcp

Configure the SIP dial peer to use the TCP transport layer protocol.

Defaults

The default for this command is that UDP is enabled.

Note The transport protocol for transport and session transport must be the same.

Command Modes

Dial-peer configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Usage Guidelines

Use show sip-ua status to ensure that the transport protocol that you set in session transport matches the protocol set in (config-sip-ua) # transport.

Examples
Router (config)# dial-peer voice 102 voip
Router (dial-peer-config)# session transport udp 
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation.

show sip-ua

To display statistics for SIP retires, timers, and current listener status, enter the show sip-ua command.

show sip-ua {retry | status | timers}
Syntax Description

retry

Displays SIP protocol retry counts.

status

Displays SIP UA listener status.

timers

Displays SIP protocol timers.

Defaults

There are no default behaviors or values for this command.

Command Modes

EXEC

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples

The following example displays output for the show sip-ua retry command:
router#show sip-ua retry
SIP UA Retry Values
invite retry count = 2
response retry count = 2
bye retry count    = 2
cancel retry count   = 1
The following example displays output for the show sip-ua status command:
router#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
The following example displays output for the show sip-ua timers command:
router#show sip-ua timers
SIP UA Timer Values
invite-wait-100 = 500   millisec 
invite-wait-180 = 30000 millisec
invite-wait-200 = 60000 millisec    
200-wait-ack = 1000  millisec
bye-wait-200    = 500   millisec
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

sip-server

To configure the SIP server interface, use the sip-server command in SIP user-agent configuration mode. This command eliminates the need to repeatedly enter the SIP server interface in the dial peers.

sip-server {dns:host-name | ipv4:ip-address[:port-number]}
Syntax Description

dns

Sets the global SIP server interface to a DNS.

host-name

A valid DNS host name takes the following format:

gateway.company.com.

ipv4:ip-address

Sets the global SIP server interface to an IP address. A valid IP address takes the following format:

xxx.xxx.xxx.xxx

port-number

(Optional) Specifies the port number for the SIP server.

Defaults

The default for this command is a null value.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# sip-ua
Router (config-sip-ua)# sip-server dns:UA-1-f0.sip.com
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

sip-ua

To enable the SIP user-agent configuration commands, with which you configure the user agent, use 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

There are no default behaviors or values for this command.

Command Modes

Global configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Usage Guidelines

Enter the sip-ua command to enter the SIP user agent-configuration sub-mode. The submode configuration commands are:

Command

Description

exit sip-ua

Exits the SIP user agent configuration mode.

inband-alerting

Specifies an inband-alerting SIP header.

retry

Configures the SIP signaling timers for retry attempts.

sip-server

Configures a SIP server interface.

timers

Configures the SIP signaling timers configuration.

transport

Enables or disables a SIP user agent transport for TCP or UDP, the protocol SIP user agents will be listening for on port 5060 (default).

Examples
Router (config)# sip-ua 
Router (config-sip-ua)# retry invite 2
Router (config-sip-ua)# retry response 2
Router (config-sip-ua)# retry bye 2
Router (config-sip-ua)# retry cancel 2
Router (config-sip-ua)# sip-server ipv4:10.0.2.254
Router (config-sip-ua)# timers invite-wait-100 500 
Router (config-sip-ua)# exit 
Related Commands

Command

Description

exit sip-ua

Exits the SIP user agent configuration.

inband-alerting

Specifies an inband-alerting SIP header.

retry

Configures the retry attempts for SIP messages.

show sip-ua

Displays statistics for SIP retires, timers, and current listener status.

sip-server

Configures the SIP server interface.

timers

Configures the SIP signaling timers.

transport

Configures the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.

timers

To configure the SIP signaling timers, use the timers command in SIP user-agent configuration mode. To reset this command to the default value, use the no form of this command.

timers {invite-wait-100 number| invite-wait-180 number | invite-wait-200 number | 200-wait-ack number | bye-wait-200 number}
no timers {invite-wait-100 number| invite-wait-180 number | invite-wait-200 number | 200-wait-ack number | bye-wait-200 number}
Syntax Description

invite-wait-100 number

Time (in milliseconds) to wait for a 100 response to an INVITE request; default = 500.

invite-wait-180 number

Time (in milliseconds) to wait for a 180 response to an INVITE request; default = 30000.

invite-wait-200 number

Time (in milliseconds) to wait for a 200 response to an INVITE request; default = 60000.

200-wait-ack number

Time (in milliseconds) to wait for a 200 response to an ACK request; default = 500.

bye-wait-200 number

Time (in milliseconds) to wait for a 200 response to a BYE request; default = 500.

Defaults

The default is the default value for each argument as listed in the Syntax Description table.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# sip-ua
Router (config-sip-ua)# timers invite-wait-100 500 
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

transport

To configure the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket, use the transport command in SIP user-agent configuration mode. This command controls whether messages reach the SIP service provider interface (SPI). By setting udp or tcp as the protocol, this will be the protocol SIP user agents will be listening for on port 5060 (default). To block reception of SIP signaling messages on a specific 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, sip-ua enables both UDP and TCP transport protocols.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
Router (config)# sip-ua
Router (config-sip-ua)# no transport tcp
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

Debug Commands

This section documents new and modified debug commands associated with the SIP feature. All other commands used with this feature are documented in the Cisco IOS Release 12.1(1)T command references. All debug commands are EXEC commands.

• debug ccsip all

• debug ccsip calls

• debug ccsip error

• debug ccsip events

• debug ccsip messages

• debug ccsip states

debug ccsip all

To enable all SIP-related debugging, enter the debug ccsip all command. To disable all debugging output, use the no form of this command.

debug ccsip all
Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Usage Guidelines

The debug ccsip all command enables the following debug SIP commands:

Command

Description

debug ccsip calls

Shows all SIP Service Provider Interface (SPI) call tracing.

debug ccsip error

Shows SIP Service Provider Interface (SPI) errors.

debug ccsip events

Shows all SIP Service Provider Interface (SPI) events tracing.

debug ccsip messages

Shows all SIP Service Provider Interface (SPI) message tracing.

debug ccsip states

Shows all SIP Service Provider Interface (SPI) state tracing.

Examples
UA-1#deb ccsip all
All SIP call tracing enabled
UA-1#
*Jan 2 18:36:38:%ISDN-6-LAYER2UP:Layer 2 for Interface Se3:23, TEI 0 changed to up
*Jan 2 18:36:49.302:0x621FA630 :State change from (STATE_NONE, SUBSTATE_NONE) to 
(STATE_IDLE, SUBSTATE_NONE)
*Jan 2 18:36:49.302: Queued event from SIP SPI :SIPSPI_EV_CC_CALL_SETUP
*Jan 2 18:36:49.302:CCSIP-SPI-CONTROL: act_idle_call_setup
*Jan 2 18:36:49.302: act_idle_call_setup:Not using Voice Class Codec
*Jan 2 18:36:49.302:act_idle_call_setup:preferred_codec set[0] type 
:g711ulaw bytes:160 
*Jan 2 18:36:49.302: Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION
*Jan 2 18:36:49.306:0x621FA630 :State change from (STATE_IDLE, SUBSTATE_NONE) to 
(STATE_IDLE, SUBSTATE_CONNECTING)
*Jan 2 18:36:49.306:0x621FA630 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to 
(STATE_IDLE, SUBSTATE_CONNECTING)
*Jan 2 18:36:49.306:CCSIP-SPI-CONTROL: act_idle_connection_created
*Jan 2 18:36:49.306:CCSIP-SPI-CONTROL: act_idle_connection_created:Connid(1) created to 
2.0.0.2:5060, local_port 6932
*Jan 2 18:36:49.310: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
*Jan 2 18:36:49.310:0x621FA630 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to 
(STATE_SENT_INVITE, SUBSTATE_NONE)
*Jan 2 18:36:49.310:
Send:
INVITE sip:9605550001@10.0.0.2;user=phone SIP/2.0
Via:SIP/2.0/UDP 10.0.0.1:6932
From:sip:9505550001@10.0.0.1
To:<sip:9605550001@10.0.0.2;user=phone>
Date:Sun 02 Jan 2000 14:36:49 EDT
Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2
Cisco-Guid:3398778967-3070345246-0-153414444
Require:com.cisco.inband-alerting
User-Agent:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq:100 INVITE
Content-Type:application/sdp
Content-Length:113
v=0
o=CiscoSystemsSIPUserAgent 6659 1152 IN IP4 10.0.0.1
s=SIP Call
c=IN IP4 10.0.0.1
m=audio 20910 RTP/AVP 0
*Jan 2 18:36:49.318:Received :
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 10.0.0.1:6932
From:sip:9505550001@10.0.0.1
To:<sip:9605550001@10.0.0.2;user=phone>
Date:Sun 02 Jan 2000 14:36:46 EDT
Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2
Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq:100 INVITE
Content-Length:0
*Jan 2 18:36:49.318:HandleUdpSocketReads :Msg enqueued for SPI with 
IPaddr:10.0.0.2:5060
*Jan 2 18:36:49.318:CCSIP-SPI-CONTROL: act_sentinvite_new_message
*Jan 2 18:36:49.318:CCSIP-SPI-CONTROL: sipSPICheckResponse
*Jan 2 18:36:49.318:0x621FA630 :State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to 
(STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
*Jan 2 18:36:49.362:Received:
SIP/2.0 180 Ringing
Via:SIP/2.0/UDP 10.0.0.1:6932
From:sip:9505550001@10.0.0.1
To:<sip:9605550001@10.0.0.2;user=phone>
Date:Sun 02 Jan 2000 14:36:46 EDT
Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2
Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Content-Type:application/sdp
CSeq:100 INVITE
Content-Length:113
v=0
o=CiscoSystemsSIPUserAgent 7548 9300 IN IP4 10.0.0.2
s=SIP Call
c=IN IP4 10.0.0.2
m=audio 20234 RTP/AVP 0
*Jan 2 18:36:49.362:HandleUdpSocketReads:Msg enqueued for SPI with IPaddr:10.0.0.2:5060
*Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: act_recdproc_new_message
*Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: sipSPICheckResponse
*Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: sipSPICheckResponse:Updating session description
*Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: act_recdproc_new_message:SDP 
MediaTypes negotiation successful!
Negotiated Codec  :g711ulaw , bytes :160
Inband Alerting  :2 
*Jan 2 18:36:49.366:0x621FA630 :State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING)
*Jan 2 18:36:49.366:CCSIP-SPI-CONTROL: ccsip_caps_ind
*Jan 2 18:36:49.366:ccsip_caps_ind:Load DSP with codec (5) g711ulaw, Bytes=160
*Jan 2 18:36:49.366:ccsip_caps_ind:set DSP for dtmf-relay = 
CC_CAP_DTMF_RELAY_INBAND_VOICE
*Jan 2 18:36:49.366:CCSIP-SPI-CONTROL: ccsip_caps_ack
*Jan 2 18:36:49.782:Received:
SIP/2.0 200 OK
Via:SIP/2.0/UDP 10.0.0.1:6932
From:sip:9505550001@10.0.0.1
To:<sip:9605550001@10.0.0.2;user=phone>
Date:Sun 02 Jan 2000 14:36:46 EDT
Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2
Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Content-Type:application/sdp
CSeq:100 INVITE
Content-Length:113
v=0
o=CiscoSystemsSIPUserAgent 6822 5961 IN IP4 10.0.0.2
s=SIP Call
c=IN IP4 10.0.0.2
m=audio 20234 RTP/AVP 0
*Jan 2 18:36:49.786:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:10.0.0.2:5060
*Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: act_recdproc_new_message
*Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: sipSPICheckResponse
*Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: sipSPICheckResponse:Updating session description
*Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: act_recdproc_new_message:SDP 
MediaTypes negotiation successful!
Negotiated Codec :g711ulaw, bytes:160
*Jan 2 18:36:49.786: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
*Jan 2 18:36:49.786:0x621FA630 :State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_ALERTING) to (STATE_ACTIVE, SUBSTATE_NONE)
*Jan 2 18:36:49.786:The Call Setup Information is:
Call Control Block (CCB) :0x621FA630
State of The Call  :STATE_ACTIVE
TCP Sockets Used   :NO
Calling Number   :9505550001
Called Number   :9605550001
Negotiated Codec   :g711ulaw
Source IP Address (Media):10.0.0.1
Source IP Port (Media):20910
Destn IP Address (Media):10.0.0.2
Destn IP Port (Media):20234
Destn SIP Addr (Control) :10.0.0.2
Destn SIP Port (Control) :5060
Destination Name   :10.0.0.2
*Jan 2 18:36:49.790:
Send:
ACK sip:9605550001@10.0.0.2;user=phone SIP/2.0
Via:SIP/2.0/UDP 10.0.0.1:6932
From:sip:9505550001@10.0.0.1
To:<sip:9605550001@10.0.0.2;user=phone>
Date:Sun 02 Jan 2000 14:36:49 EDT
Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2
Content-Type:application/sdp
Content-Length:113
CSeq:100 ACK
v=0
o=CiscoSystemsSIPUserAgent 8267 3722 IN IP4 10.0.0.1
s=SIP Call
c=IN IP4 10.0.0.1
m=audio 20910 RTP/AVP 0
*Jan 2 18:37:20.893: Queued event From SIP SPI to CCAPI/DNS:SIPSPI_EV_CC_CALL_DISCONNECT
*Jan 2 18:37:20.893:CCSIP-SPI-CONTROL: act_active_disconnect
*Jan 2 18:37:20.893: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
*Jan 2 18:37:20.893:0x621FA630 :State change from (STATE_ACTIVE, SUBSTATE_NONE) to 
(STATE_DISCONNECTING, SUBSTATE_NONE)
*Jan 2 18:37:20.897:
Send:
BYE <sip:9605550001@10.0.0.2;user=phone> SIP/2.0
Via:SIP/2.0/UDP 10.0.0.1:6932
From:sip:9505550001@10.0.0.1
To:<sip:9605550001@10.0.0.2;user=phone>
Date:Sun 02 Jan 2000 14:36:49 EDT
Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2
User-Agent:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq:101 BYE
Content-Length:0
*Jan 2 18:37:20.901:Received:
SIP/2.0 200 OK
Via:SIP/2.0/UDP 10.0.0.1:6932
From:sip:9505550001@10.0.0.1
To:<sip:9605550001@10.0.0.2;user=phone>
Date:Sun 02 Jan 2000 14:37:18 EDT
Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2
Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Content-Length:0
CSeq:101 BYE
*Jan 2 18:37:20.901:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:10.0.0.2:5060
*Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: act_disconnecting_new_message
*Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response
*Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: sipSPICheckResponse
*Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: sipSPICallCleanup
*Jan 2 18:37:20.901: Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTION
*Jan 2 18:37:20.905:0x621FA630 :State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to 
(STATE_DEAD, SUBSTATE_NONE)
*Jan 2 18:37:20.905:The Call Setup Information is :
Call Control Block (CCB) :0x621FA630
State of The Call  :STATE_DEAD
TCP Sockets Used   :NO
Calling Number   :9505550001
Called Number   :9605550001
Negotiated Codec   :g711ulaw
Source IP Address (Media):10.0.0.1
Source IP Port (Media):20910
Destn IP Address (Media):10.0.0.2
Destn IP Port (Media):20234
Destn SIP Addr (Control) :10.0.0.2
Destn SIP Port (Control) :5060
Destination Name   :10.0.0.2
*Jan 2 18:37:20.905:  Disconnect Cause (CC) :16
Disconnect Cause (SIP) :200
debug ccsip calls

To show all SIP Service Provider Interface (SPI) call tracing, enter the debug ccsip calls command. This command traces the SIP call details as updated in the SIP call control block.

debug ccsip calls
Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
UA-1#deb ccsip calls
SIP Call statistics tracing is enabled
UA-1#
*Jan  2 18:21:01.245: The Call Setup Information is:
        Call Control Block (CCB): 0x62202E3C
        State of The Call       : STATE_ACTIVE
         TCP Sockets Used        : NO
         Calling Number          : 9505550001
         Called Number           : 9605550001
         Negotiated Codec        : g711ulaw
         Source IP Address (Media): 10.0.0.1
         Source IP Port    (Media): 20754
         Destn  IP Address (Media): 10.0.0.2
         Destn  IP Port    (Media): 20748
         Destn SIP Addr (Control): 10.0.0.2
         Destn SIP Port (Control): 5060
         Destination Name        : 10.0.0.2
*Jan  2 18:21:32.708: The Call Setup Information is :
        Call Control Block (CCB): 0x62202E3C
        State of The Call       : STATE_DEAD
         TCP Sockets Used        : NO
         Calling Number          : 9505550001
         Called Number           : 9605550001
         Negotiated Codec        : g711ulaw
         Source IP Address (Media): 10.0.0.1
         Source IP Port    (Media): 20754
         Destn  IP Address (Media): 10.0.0.2
         Destn  IP Port    (Media): 20748
         Destn SIP Addr (Control): 10.0.0.2
         Destn SIP Port (Control): 5060
         Destination Name        : 10.0.0.2
*Jan  2 18:21:32.708:         Disconnect Cause (CC)   : 16
        Disconnect Cause (SIP)  : 200
debug ccsip error

To show SIP Service Provider Interface (SPI) errors, enter the debug ccsip error command. This command traces all error messages generated from errors encountered by the SIP subsystem.

debug ccsip error
Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
UA-1#deb ccsip error
SIP Call error tracing is enabled
UA-1#
*Jan  2 18:24:25.281: CCSIP-SPI-CONTROL:  act_idle_call_setup
*Jan  2 18:24:25.281:  act_idle_call_setup:Not using Voice Class Codec
*Jan  2 18:24:25.281: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 
160 
*Jan  2 18:24:25.281: CCSIP-SPI-CONTROL:  act_idle_connection_created
*Jan  2 18:24:25.285: CCSIP-SPI-CONTROL:  act_idle_connection_created: Connid(1) created 
to 2.0.0.2:5060, local_port 9830
*Jan  2 18:24:25.293: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
2.0.0.2:5060
*Jan  2 18:24:25.293: CCSIP-SPI-CONTROL:  act_sentinvite_new_message
*Jan  2 18:24:25.293: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Jan  2 18:24:25.337: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
2.0.0.2:5060
*Jan  2 18:24:25.341: CCSIP-SPI-CONTROL:  act_recdproc_new_message
*Jan  2 18:24:25.341: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Jan  2 18:24:25.341: CCSIP-SPI-CONTROL:  sipSPICheckResponse : Updating session 
description
*Jan  2 18:24:25.341: CCSIP-SPI-CONTROL:  act_recdproc_new_message: SDP 
MediaTypes negotiation successful!
Negotiated Codec      : g711ulaw , bytes :160
Inband Alerting       : 2 
*Jan  2 18:24:25.341: CCSIP-SPI-CONTROL:  ccsip_caps_ind
*Jan  2 18:24:25.341: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160
*Jan  2 18:24:25.341: ccsip_caps_ind: set DSP for dtmf-relay = 
CC_CAP_DTMF_RELAY_INBAND_VOICE
*Jan  2 18:24:25.341: CCSIP-SPI-CONTROL:  ccsip_caps_ack
*Jan  2 18:24:25.769: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
2.0.0.2:5060
*Jan  2 18:24:25.773: CCSIP-SPI-CONTROL:  act_recdproc_new_message
*Jan  2 18:24:25.773: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Jan  2 18:24:25.773: CCSIP-SPI-CONTROL:  sipSPICheckResponse : Updating session 
description
*Jan  2 18:24:25.773: CCSIP-SPI-CONTROL:  act_recdproc_new_message: SDP 
MediaTypes negotiation successful!
Negotiated Codec      : g711ulaw , bytes :160
*Jan  2 18:24:57.012: CCSIP-SPI-CONTROL:  act_active_disconnect
*Jan  2 18:24:57.020: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
2.0.0.2:5060
*Jan  2 18:24:57.020: CCSIP-SPI-CONTROL:  act_disconnecting_new_message
*Jan  2 18:24:57.020: CCSIP-SPI-CONTROL:  sact_disconnecting_new_message_response
*Jan  2 18:24:57.020: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Jan  2 18:24:57.020: CCSIP-SPI-CONTROL:  sipSPICallCleanup
debug ccsip events

To show all SIP Service Provider Interface (SPI) events tracing, enter the debug ccsip events command. This command traces the events posted to SIP SPI from all interfaces.

debug ccsip events
Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
UA-1#debug ccsip events
SIP Call events tracing is enabled
UA-1#
*Jan 2 18:28:06.784: Queued event from SIP SPI :SIPSPI_EV_CC_CALL_SETUP
*Jan 2 18:28:06.784: Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION
*Jan 2 18:28:06.792: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
*Jan 2 18:28:07.284: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
*Jan 2 18:28:38.384: Queued event From SIP SPI to CCAPI/DNS:SIPSPI_EV_CC_CALL_DISCONNECT
*Jan 2 18:28:38.388: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
debug ccsip messages

To show all SIP Service Provider Interface (SPI) message tracing, enter the debug ccsip messages command. This command traces the SIP messages exchanged between the SIP user agent client (UAC) and the access server.

debug ccsip messages
Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
UA-1#deb ccsip message
SIP Call messages tracing is enabled
UA-1#
*Jan  2 20:40:40.937: 
Send:
INVITE sip:9605550001@10.0.0.2;user=phone SIP/2.0
Via: SIP/2.0/UDP  10.0.0.1:2537
From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1>
To: <sip:9605550001@10.0.0.2;user=phone>
Date: Sun 02 Jan 2000 16:40:40 EDT
Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2
Cisco-Guid: 3398778967-3070345273-0-160846216
Require: com.cisco.inband-alerting
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 100 INVITE
Content-Type: application/sdp
Content-Length: 113
v=0
o=CiscoSystemsSIPUserAgent 9074 8380 IN IP4 10.0.0.1
s=SIP Call
c=IN IP4 10.0.0.1
m=audio 20610 RTP/AVP 0
*Jan  2 20:40:40.945: Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  10.0.0.1:2537
From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1>
To: <sip:9605550001@10.0.0.2;user=phone>
Date: Sun 02 Jan 2000 16:40:38 EDT
Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 100 INVITE
Content-Length: 0
*Jan  2 20:40:40.993: Received: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP  10.0.0.1:2537
From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1>
To: <sip:9605550001@10.0.0.2;user=phone>
Date: Sun 02 Jan 2000 16:40:38 EDT
Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Content-Type: application/sdp
CSeq: 100 INVITE
Content-Length: 113
v=0
o=CiscoSystemsSIPUserAgent 6706 3098 IN IP4 10.0.0.2
s=SIP Call
c=IN IP4 10.0.0.2
m=audio 20460 RTP/AVP 0
*Jan  2 20:40:41.421: Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  10.0.0.1:2537
From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1>
To: <sip:9605550001@10.0.0.2;user=phone>
Date: Sun 02 Jan 2000 16:40:38 EDT
Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Content-Type: application/sdp
CSeq: 100 INVITE
Content-Length: 113
v=0
o=CiscoSystemsSIPUserAgent 7641 2845 IN IP4 10.0.0.2
s=SIP Call
c=IN IP4 10.0.0.2
m=audio 20460 RTP/AVP 0
*Jan  2 20:40:41.425: 
Send:
ACK sip:9605550001@10.0.0.2;user=phone SIP/2.0
Via: SIP/2.0/UDP  10.0.0.1:2537
From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1>
To: <sip:9605550001@10.0.0.2;user=phone>
Date: Sun 02 Jan 2000 16:40:40 EDT
Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2
Content-Type: application/sdp
Content-Length: 113
CSeq: 100 ACK
v=0
o=CiscoSystemsSIPUserAgent 6286 8863 IN IP4 10.0.0.1
s=SIP Call
c=IN IP4 10.0.0.1
m=audio 20610 RTP/AVP 0
*Jan  2 20:41:12.596: 
Send:
BYE <sip:9605550001@10.0.0.2;user=phone> SIP/2.0
Via: SIP/2.0/UDP  10.0.0.1:2537
From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1>
To: <sip:9605550001@10.0.0.2;user=phone>
Date: Sun 02 Jan 2000 16:40:40 EDT
Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 BYE
Content-Length: 0
*Jan  2 20:41:12.600: Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  10.0.0.1:2537
From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1>
To: <sip:9605550001@10.0.0.2;user=phone>
Date: Sun 02 Jan 2000 16:41:09 EDT
Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Content-Length: 0
CSeq: 101 BYE
debug ccsip states

To show all SIP Service Provider Interface (SPI) state tracing, enter the debug ccsip states command. This command traces the state machine changes of SIP SPI and displays the state transitions.

debug ccsip states
Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the AS5400 and AS5350 platforms.

12.2(2)XB1

This command was implemented on the Cisco AS5850 platform.

Examples
UA-1#deb ccsip states
SIP Call states tracing is enabled
UA-1#
*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_CONNECTION
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. SS7 (signaling system 7) feature in which a series of digits, either analog or digital, are included in the call, identifying the telephone number of the calling device. In other words, ANI identifies the number of the calling party.

CAS—channel associated signaling.

CCAPI—call control applications programming interface.

CLI—command line interface. Interface that allows the user to interact with the operating system by entering commands and optional arguments. The UNIX operating system and DOS provide CLIs.

CO—central office. Local telephone company office to which all local loops in a given area connect and in which circuit switching of subscriber lines occurs.

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 (VoIP), there are two types of dial peers: POTS and VoIP.

DNS—Domain Name System. System used in the Internet for translating names of network nodes into addresses.

DNIS—dialed number identification service (the called number).

DSP—digital signal processor.

DTMF—dual tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch tone).

E.164—The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.

E&M—recEive and transMit (or ear and mouth). Trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco's analog E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines). E&M is also available on E1 and T1 digital interfaces.

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.

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. Local or regional telephone company that owns and operates a telephone network and the customer lines that connect to it.

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—Multifrequency 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—An 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.

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.

PSTN—Public switched telephone network. PSTN refers to the local telephone company.

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.

SIP—Session Initiation Protocol. This is a protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999.

SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.

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.

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.

Hierarchical Navigation

Cisco IOS Software Releases 12.1 T

Session Initiation Protocol for VoIP on Cisco Access Platforms

Downloads

Table Of Contents

Session Initiation Protocol for Voice over IP on Cisco Access Platforms

Feature Overview

Benefits

Restrictions

Related Features and Technologies

DNS SRV

Related Documents

Supported Platforms

Supported Standards, MIBs, and RFCs

Standards

MIBs

RFCs

Prerequisites

Configuration Tasks

Configuring the SIP User Agent (UA)

Configuring a VoIP Dial Peer

Configuring a POTS Dial Peer

Verifying the SIP Feature Configuration

Configuration Examples

Basic SIP Configuration

Configuring SIP with Multiple Codecs

Command Reference

default

Syntax Description

Defaults

Command Modes

Command History

Examples

Related Commands

exit sip-ua

Syntax Description

Defaults

Command Modes

Command History

Examples

Related Commands

gw-accounting

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

inband-alerting

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

max-redirects

Syntax Description

Defaults

Command Modes

Command History

Examples

Related Commands

retry

Syntax Description

Defaults

Command Modes

Command History

Examples

Related Commands

session protocol

Syntax Description

Defaults

Command Modes

Command History

Examples

Related Commands

session target

Syntax Description