Basic SIP Configuration

This chapter provides basic configuration information for the following features:

SIP Register Support SIP Redirect Processing Enhancement SIP 300 Multiple Choice Messages SIP implementation enhancements:
Interaction with Forking Proxies SIP Intra-Gateway Hairpinning

Feature History for SIP Register Support, SIP Redirect Processing Enhancement, and SIP 300 Multiple Choice Messages

Release 12.2(15)ZJ 12.3(4)T

Modification This feature was introduced. This feature was integrated into the release.

Feature History for SIP Implementation Enhancements: Interaction with Forking Proxies and SIP Intra-Gateway Hairpinning

Release 12.2(2)XB 12.2(8)T

Modification These features were introduced. This feature were integrated into the release.

Finding Support Information for Platforms and Cisco IOS Software Images

Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.

Americas Headquarters: Cisco Systems, Inc., 170 West Tasman Drive, San Jose, CA 95134-1706 USA

2007 Cisco Systems, Inc. All rights reserved.

Basic SIP Configuration Contents

Contents

Prerequisites for Basic SIP Configuration, page 40 Information About Basic SIP Configuration, page 40 How to Perform Basic SIP Configuration, page 42 Configuration Examples for Basic SIP Configuration, page 58 Additional References, page 66

Prerequisites for Basic SIP Configuration

SIP Redirect Processing Enhancement Feature

Ensure that your SIP gateway supports 300 or 302 Redirect messages.

Information About Basic SIP Configuration

To perform basic SIP configuration tasks, you should understand the following concepts:

SIP Register Support, page 40 SIP Redirect Processing Enhancement, page 40 Sending SIP 300 Multiple Choice Messages, page 41

SIP Register Support

With H.323, Cisco IOS gateways can register E.164 numbers of a POTS dial peer with a gatekeeper, which informs the gatekeeper of a users contact information. SIP gateways allow the same functionality, but with the registration taking place with a SIP proxy or registrar. SIP gateways allow registration of E.164 numbers to a SIP proxy or registrar on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and local SCCP phones. When registering dial peers with an external registrar, you can also register with a secondary SIP proxy or registrar to provide redundancy. The secondary registration can be used if the primary registrar fails. SIP gateways allow registration of E.164 numbers to a SIP proxy or registrar server on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and local SCCP phones. By default, SIP gateways do not generate SIP Register messages. The following tasks set up the gateway to register E.164 telephone numbers with an external SIP registrar.

Note

There are no commands that allow registration between the H.323 and SIP protocols.

SIP Redirect Processing Enhancement

SIP Redirect Processing allows flexibility in the handling of incoming redirect or 3xx class of responses. Redirect responses can be enabled or disabled through the command-line interface, providing a benefit to service providers who deploy Cisco SIP gateways. Redirect processing is active by default, which

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means that SIP gateways handle incoming 3xx messages in compliance with RFC 2543. RFC 2543 states that redirect response messages are used by SIP user agents to initiate a new Invite when a user agent learns that a user has moved from a previously known location. In accordance with RFC 2543-bis-04, the processing of 3xx redirection is as follows:

The uniform resource identifier (URI) of the redirected INVITE is updated to contain the new contact information provided by the 3xx redirect message. The transmitted CSeq number found in the CSeq header is increased by one. The new INVITE includes the updated CSeq. The To, From, and Call ID headers that identify the call leg remain the same. The same Call ID gives consistency when capturing billing history. The UAC retries the request at the new address given by the 3xx Contact header field.

Redirect handling can be disabled by using the no redirection command in SIP user-agent configuration mode. In this case, the user agent treats incoming 3xx responses as 4xx error class responses. The call is not redirected, and is instead released with the appropriate PSTN cause-code message. Table 1 shows the mapping of 3xx responses to 4xx responses.
Table 1 Mapping of 3xx Responses to 4xx Responses

Redirection (3xx) Response Message 300 Multiple choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service <any other 3xx response>

Mapping to 4xx (Client Error) Response 410 Gone 410 Gone 480 Temporarily Unavailable 410 Gone 410 Gone 410 Gone

SIP Redirect Processing generates call history information with appropriate release cause codes that maybe used for accounting or statistics purposes. When a 3xx response is mapped to 4xx class of response, the cause code stored in call history is based on the mapped 4xx response code. Call redirection must be enabled on the gateway for SIP call transfer involving redirect servers to be successful. The Cisco IOS voice gateway can also use call redirection if an incoming VoIP call matches an outbound VoIP dial peer. The gateway sends a 300 or 302 Redirect message to the call originator, allowing the originator to reestablish the call. Two commands allow you to enable the redirect functionality, globally or on a specific inbound dial peer: redirect ip2ip (dial-peer) and redirect ip2ip (voice service).

Sending SIP 300 Multiple Choice Messages

Originally, when a call was redirected, the SIP gateway would send a 302 Moved Temporarily message. The first longest match route on a gateway (dial-peer destination pattern) was used in the Contact header of the 302 message. Now, if multiple routes to a destination exist for a redirected number (multiple dial peers are matched), the SIP gateway sends a 300 Multiple Choice message, and the multiple routes in the Contact header are listed. The redirect contact order command gives you the flexibility to choose the order in which routes appear in the Contact header.

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How to Perform Basic SIP Configuration

This section contains the following procedures:

Configuring SIP VoIP Services on a Cisco Gateway, page 42 Configuring SIP Register Support, page 44 Configuring SIP Redirect Processing Enhancement, page 46 Configuring SIP 300 Multiple Choice Messages, page 49 Configuring SIP Implementation Enhancements, page 50
Interaction with Forking Proxies, page 51 SIP Intra-Gateway Hairpinning, page 51

Verifying SIP Gateway Status, page 52

Note

For help with a procedure, see the verification and troubleshooting sections listed above.

Configuring SIP VoIP Services on a Cisco Gateway

This section contains the following procedures:

Shut Down or Enable VoIP Service on Cisco Gateways, page 42 Shut Down or Enable VoIP Submodes on Cisco Gateways, page 43

Shut Down or Enable VoIP Service on Cisco Gateways

To shut down or enable VoIP service on Cisco gateways, perform the following steps.

SUMMARY STEPS
1. 2. 3. 4. 5.

enable configure terminal voice service voip [no] shutdown exit

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DETAILED STEPS

Command or Action
Step 1
enable

Purpose Enters privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted. Enters global configuration mode.

Example:
Router> enable

Step 2

configure terminal

Example:
Router# configure terminal

Step 3

voice service voip

Enters voice-service VoIP configuration mode.

Example:
Router(config)# voice service voip

Step 4

[no] shutdown [forced]

Shuts down or enables VoIP call services.

Example:
Router(config-voi-serv)# shutdown forced

Step 5

exit

Exits the current mode.

Example:
Router(config-voi-serv)# exit

Shut Down or Enable VoIP Submodes on Cisco Gateways

To shut down or enable VoIP submodes on Cisco gateways, perform the following steps.

SUMMARY STEPS
1. 2. 3. 4. 5. 6.

enable configure terminal voice service voip sip [no] call service stop exit

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DETAILED STEPS

Command or Action
Step 1
enable

Purpose Enters privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted. Enters global configuration mode.

Example:
Router> enable

Step 2

configure terminal

Example:
Router# configure terminal

Step 3

voice service voip

Enters voice-service VoIP configuration mode.

Example:
Router(config)# voice service voip

Step 4

sip

Enters SIP configuration mode.

Example:
Router(config-voi-serv)# sip

Step 5

[no] call service stop [forced] [maintain-registration]

Shuts down or enables VoIP call services for the selected submode.

Example:
Router(conf-serv-sip)# call service stop maintain-registration

Step 6

exit

Exits the current mode.

Example:
Router(conf-serv-sip)# exit

Configuring SIP Register Support

To configure SIP register support, perform the following steps.

SUMMARY STEPS
1. 2. 3. 4. 5. 6. 7.

enable configure terminal sip-ua registrar retry register timers register exit

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DETAILED STEPS

Command or Action
Step 1
enable

Purpose Enters privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted. Enters global configuration mode.

Example:
Router> enable

Step 2

configure terminal

Example:
Router# configure terminal

Step 3

sip-ua

Enters SIP user-agent configuration mode.

Example:
Router(config)# sip-ua

Step 4

registrar {dns:address | ipv4:destination-address} expires seconds [tcp] [secondary]

Registers E.164 numbers on behalf of analog telephone voice ports (FXS) and IP phone virtual voice ports (EFXS) with an external SIP proxy or SIP registrar server. Keywords and arguments are as follows:

Example:
Router(config-sip-ua)# registrar ipv4:10.8.17.40 expires 3600 secondary

dns:addressDomain-name server that resolves the name of the dial peer to receive calls. ipv4:destination-addressIP address of the dial peer to receive calls. expires secondsDefault registration time, in seconds. tcpSets transport layer protocol to TCP. UDP is the default. secondarySpecifies registration with a secondary SIP proxy or registrar for redundancy purposes. Optional.

Step 5

retry register number

Example:
Router(config-sip-ua)# retry register 10

Use this command to set the total number of SIP Register messages that the gateway should send. The argument is as follows:

numberNumber of Register message retries. Range: 1 to 10. Default: 10.

Step 6

timers register milliseconds

Example:
Router(config-sip-ua)# timers register 500

Use this command to set how long the SIP user agent waits before sending register requests. The argument is as follows:

millisecondsWaiting time, in ms. Range: 100 to 1000. Default: 500.

Step 7

exit

Exits the current mode.

Example:
Router(config-sip-ua)# exit

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Configuring SIP Redirect Processing Enhancement

This section provides the following information:

Configure Call-Redirect Processing Enhancement, page 46 Configuring SIP 300 Multiple Choice Messages, page 49

Configure Call-Redirect Processing Enhancement

Redirect processing using the redirection command is enabled by default. To disable and then reset redirect processing, perform the steps listed in this section:

Configuring Call-Redirect Processing Enhancement, page 46

IP-to-IP call redirection can be enabled globally or on a dial-peer basis. To configure, perform the steps listed in these sections:

Configuring Call Redirect to Support Calls Globally, page 47 Configuring Call Redirect to Support Calls on a Specific VoIP Dial Peer, page 48

Configuring Call-Redirect Processing Enhancement

To configure call-redirect processing enhancement, perform the following steps.

SUMMARY STEPS
1. 2. 3. 4. 5. 6.

enable configure terminal sip-ua no redirection redirection exit

DETAILED STEPS

Command or Action
Step 1
enable

Purpose Enters privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted. Enters global configuration mode.

Example:
Router> enable

Step 2

configure terminal

Example:
Router# configure terminal

Step 3

sip-ua

Enters SIP user-agent configuration mode.

Example:
Router(config)# sip-ua

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Command or Action
Step 4
no redirection

Purpose Disables redirect handlingcauses the gateway to treat incoming 3xx responses as 4xx error class responses.

Example:
Router(config-sip-ua)# no redirection

Step 5

redirection

Example:
Router(config-sip-ua)# redirection

Resets call redirection to work as specified in RFC 2543. The command default redirection also resets call redirection to work as specified in RFC 2543. Exits the current mode.

Step 6

exit

Example:
Router(config-sip-ua)# exit

Configuring Call Redirect to Support Calls Globally

To configure call redirect to support calls globally, perform the following steps.

Note

To enable global IP-to-IP call redirection for all VoIP dial peers, use voice-service configuration mode. The default SIP application supports IP-to-IP redirection.

SUMMARY STEPS
1. 2. 3. 4. 5.

enable configure terminal voice service voip redirect ip2ip exit

DETAILED STEPS

Command or Action
Step 1
enable

Purpose Enters privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted. Enters global configuration mode.

Example:
Router> enable

Step 2

configure terminal

Example:
Router# configure terminal

Step 3

voice service voip

Enters voice-service VoIP configuration mode.

Example:
Router(config)# voice service voip

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Command or Action
Step 4
redirect ip2ip

Purpose Redirect SIP phone calls to SIP phone calls globally on a gateway using the Cisco IOS voice gateway.

Example:
Router(conf-voi-serv)# redirect ip2ip

Step 5

exit

Exits the current mode.

Example:
Router(conf-voi-serv)# exit

Configuring Call Redirect to Support Calls on a Specific VoIP Dial Peer

To configure call redirect to support calls on a specific VoIP dial peer, perform the following steps.

Note

To specify IP-to-IP call redirection for a specific VoIP dial peer, configure it on an inbound dial peer in dial-peer configuration mode. The default application on SIP SRST supports IP-to-IP redirection. When IP-to-IP redirection is configured in dial-peer configuration mode, the configuration on the specific inbound dial peer takes precedence over the global configuration entered under voice service configuration.

SUMMARY STEPS
1. 2. 3. 4. 5. 6.

enable configure terminal dial-peer voice voip application redirect ip2ip exit

DETAILED STEPS

Command or Action
Step 1
enable

Purpose Enters privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted. Enters global configuration mode.

Example:
Router> enable

Step 2

configure terminal

Example:
Router# configure terminal

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Command or Action
Step 3
dial-peer voice tag voip

Purpose Use this command to enter dial-peer configuration mode. The argument is as follows:

Example:
Router(config)# dial-peer voice 29 voip

tagDigits that define a particular dial peer. Range: 1to 2,147,483,647 (enter without commas).

Step 4

application application-name

Enables a specific application on a dial peer. The argument is as follows:

Example:
Router(config-dial-peer)# application session

application-nameName of the predefined application you wish to enable on the dial peer. For SIP, the default Tcl application (from the Cisco IOS image) is session and can be applied to both VoIP and POTS dial peers. The application must support IP-to-IP redirection

Step 5

redirect ip2ip

Redirects SIP phone calls to SIP phone calls on a specific VoIP dial peer using the Cisco IOS voice gateway.

Example:
Router(conf-dial-peer)# redirect ip2ip

Step 6

exit

Exits the current mode.

Example:
Router(conf-dial-peer)# exit

Configuring SIP 300 Multiple Choice Messages

This section contains the following information:

Sending SIP 300 Multiple Choice Messages, page 41 Configuring Sending of SIP 300 Multiple Choice Messages, page 49

Configuring Sending of SIP 300 Multiple Choice Messages

To send SIP 300 Multiple Choice messages, perform the following steps.

Note

If multiple routes to a destination exist for a redirected number (multiple dial peers are matched), the SIP gateway sends a 300 Multiple Choice message and the multiple routes in the Contact header are listed. This configuration allows users to choose the order in which the routes appear in the Contact header.

SUMMARY STEPS
1. 2. 3. 4.

enable configure terminal voice service voip sip

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5. 6.

redirect contact order exit

DETAILED STEPS

Command or Action
Step 1
enable

Purpose Enters privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted. Enters global configuration mode.

Example:
Router> enable

Step 2

configure terminal

Example:
Router# configure terminal

Step 3

voice service voip

Enters voice-service VoIP configuration mode.

Example:
Router(config)# voice service voip

Step 4

sip

Enters SIP configuration mode.

Example:
Router(config-voi-serv)# sip

Step 5

redirect contact order [best-match | longest-match]

Sets the order of contacts in the 300 Multiple Choice Message. Keywords are as follows:

Example:
Router(conf-serv-sip)# redirect contact order best-match

best-matchUse the current system configuration to set the order of contacts. longest-matchSet the contact order by using the destination pattern longest match first, and then the second longest match, the third longest match, and so on. This is the default.

Step 6

exit

Exits the current mode.

Example:
Router(conf-serv-sip)# exit

Configuring SIP Implementation Enhancements

Minor underlying or minimally configurable features are described in the following sections:

Interaction with Forking Proxies, page 51 SIP Intra-Gateway Hairpinning, page 51

For additional information on SIP implementation enhancements, see Achieving SIP RFC Compliance on page 67.

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Interaction with Forking Proxies

Call forking enables the terminating gateway to handle multiple requests and the originating gateway to handle multiple provisional responses for the same call. Call forking is required for the deployment of the find me/follow me type of services. Support for call forking enables the terminating gateway to handle multiple requests and the originating gateway to handle multiple provisional responses for the same call. Interaction with forking proxies applies to gateways acting as a UAC, and takes place when a user is registered to several different locations. When the UAC sends an INVITE message to a proxy, the proxy forks the request and sends it to multiple user agents. The SIP gateway processes multiple 18X responses by treating them as independent transactions under the same call ID. When the relevant dial peers are configured for QoS, the gateway maintains state and initiates RSVP reservations for each of these independent transactions. When it receives an acknowledgment, such as a 200 OK, the gateway accepts the successful acknowledgment and destroys state for all other transactions. The forking feature sets up RSVP for each transaction only if the dial peers are configured for QoS. If not, the calls proceed as best-effort. Support for interaction with forking proxies applies only to gateways acting as UACs. It does not apply when the gateway acts as a UAS. In that case, the proxy forks multiple INVITES with the same call ID to the same gateway but with different request URLs. Also, the forking feature sets up RSVP for each transaction only if the dial peers are configured for QoS. If not, the calls proceed as best-effort.

SIP Intra-Gateway Hairpinning

SIP hairpinning is a call routing capability in which an incoming call on a specific gateway is signaled through the IP network and back out the same gateway. This can be a PSTN call routed into the IP network and back out to the PSTN over the same gateway (see Figure 11).
Figure 11 PSTN Hairpinning Example

PSTN

Gateway call id - x call id - x

IP network
37698

Similarly, SIP hairpinning can be a call signaled from a line (for example, a telephone line) to the IP network and back out to a line on the same access gateway (see Figure 12).
Figure 12 Telephone Line Hairpinning Example

Gateway Line 1 Line 2 call id - y call id - y

IP network
37699

With SIP hairpinning, unique gateways for ingress and egress are unnecessary. SIP supports plain old telephone service (POTS)-to-POTS hairpinning (which means that the call comes in one voice port and is routed out another voice port). It also supports POTS-to-IP call legs and IP-to-POTS call legs. However, it does not support IP-to-IP hairpinning. This means that the SIP gateway cannot take an inbound SIP call and reroute it back to another SIP device using the VoIP dial peers.

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Only minimal configuration is required for this feature. To enable hairpinning on the SIP gateway, see the following configuration example for dial peers. Note that:

The POTS dial peer must have preference 2 defined, and the VoIP dial peer must have preference 1 defined. This ensures that the call is sent out over IP, not Plain Old Telephone Service (POTS). The session target is the same gateway because the call is being redirected to it.

! dial-peer voice 53001 pots preference 2 destination-pattern 5300001 prefix 5300001 ! dial-peer voice 53002 pots preference 2 destination-pattern 5300002 prefix 5300002 ! dial-peer voice 530011 voip preference 1 destination-pattern 5300001 session protocol sipv2 session target ipv4:10.1.1.41 playout-delay maximum 300 codec g711alaw ! dial-peer voice 530022 voip preference 1 destination-pattern 5300002 session protocol sipv2 session target ipv4:10.1.1.41 playout-delay maximum 300 codec g711alaw

Verifying SIP Gateway Status

To verify SIP gateway status and configuration, perform the following steps as appropriate (commands are listed in alphabetical order).

SUMMARY STEPS
1. 2. 3. 4. 5.

show sip service show sip-ua register status show sip-ua statistics show sip-ua status show sip-ua timers

DETAILED STEPS
Step 1

show sip service Use this command to display the status of SIP call service on a SIP gateway. The following sample output shows that SIP call service is enabled:
Router# show sip service

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SIP Service is up

The following sample output shows that SIP call service was shut down with the shutdown command:
Router# show sip service SIP service is shut globally under 'voice service voip'

The following sample output shows that SIP call service was shut down with the call service stop command:
Router# show sip service SIP service is shut under 'voice service voip', 'sip' submode

The following sample output shows that SIP call service was shut down with the shutdown forced command:
Router# show sip service SIP service is forced shut globally under 'voice service voip'

The following sample output shows that SIP call service was shut down with the call service stop forced command:
Router# show sip service SIP service is forced shut under 'voice service voip', 'sip' submode

Step 2

show sip-ua register status Use this command to display the status of E.164 numbers that a SIP gateway has registered with an external primary SIP registrar.
Router# show sip-ua register status Line 4001 4002 5100 9998 peer expires(sec) registered 20001 596 no 20002 596 no 1 596 no 2 596 no

Step 3

show sip-ua statistics Use this command to display response, traffic, and retry SIP statistics, including whether call redirection is disabled. The following sample shows that four registers were sent:
Router# show sip-ua statistics SIP Response Statistics (Inbound/Outbound) Informational: Trying 0/0, Ringing 0/0, Forwarded 0/0, Queued 0/0, SessionProgress 0/0 Success: OkInvite 0/0, OkBye 0/0, OkCancel 0/0, OkOptions 0/0, OkPrack 0/0, OkPreconditionMet 0/0, OkSubscribe 0/0, OkNOTIFY 0/0,

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OkInfo 0/0, 202Accepted 0/0 OkRegister 12/49 Redirection (Inbound only except for MovedTemp(Inbound/Outbound)) : MultipleChoice 0, MovedPermanently 0, MovedTemporarily 0/0, UseProxy 0, AlternateService 0 Client 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, 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 0/0, RequestCancel 0/0, NotAcceptableMedia 0/0, BadEvent 0/0, SETooSmall 0/0 Server Error: InternalError 0/0, NotImplemented 0/0, BadGateway 0/0, ServiceUnavail 0/0, GatewayTimeout 0/0, BadSipVer 0/0, PreCondFailure 0/0 Global Failure: BusyEverywhere 0/0, Decline 0/0, NotExistAnywhere 0/0, NotAcceptable 0/0 Miscellaneous counters: RedirectRspMappedToClientErr 0 SIP Total Traffic Statistics (Inbound/Outbound) Invite 0/0, Ack 0/0, Bye 0/0, Cancel 0/0, Options 0/0, Prack 0/0, Comet 0/0, Subscribe 0/0, NOTIFY 0/0, Refer 0/0, Info 0/0 Register 49/16 Retry Statistics Invite 0, Bye 0, Cancel 0, Response 0, Prack 0, Comet 0, Reliable1xx 0, NOTIFY 0 Register 4 SDP application statistics: Parses: 0, Builds 0 Invalid token order: 0, Invalid param: 0 Not SDP desc: 0, No resource: 0 Last time SIP Statistics were cleared: <never>

The following sample output shows the RedirectResponseMappedToClientError status message. An incremented number indicates that 3xx responses are to be treated as 4xx responses. When call redirection is enabled (default), the RedirectResponseMappedToClientError status message is not incremented.
Router# show sip-ua statistics SIP Response Statistics (Inbound/Outbound) Informational: Trying 0/0, Ringing 0/0, Forwarded 0/0, Queued 0/0, SessionProgress 0/0 Success:

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OkInvite 0/0, OkBye 0/0, OkCancel 0/0, OkOptions 0/0, OkPrack 0/0, OkPreconditionMet 0/0, OKSubscribe 0/0, OkNotify 0/0, 202Accepted 0/0 Redirection (Inbound only): MultipleChoice 0, MovedPermanently 0, MovedTemporarily 0, UseProxy 0, AlternateService 0 Client 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, 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 0/0, RequestCancel 0/0 NotAcceptableMedia 0/0, BadEvent 0/0 Server Error: InternalError 0/0, NotImplemented 0/0, BadGateway 0/0, ServiceUnavail 0/0, GatewayTimeout 0/0, BadSipVer 0/0, PreCondFailure 0/0 Global Failure: BusyEverywhere 0/0, Decline 0/0, NotExistAnywhere 0/0, NotAcceptable 0/0 Miscellaneous counters: RedirectResponseMappedToClientError 1, SIP Total Traffic Statistics (Inbound/Outbound) Invite 0/0, Ack 0/0, Bye 0/0, Cancel 0/0, Options 0/0, Prack 0/0, Comet 0/0, Subscribe 0/0, Notify 0/0, Refer 0/0 Retry Statistics Invite 0, Bye 0, Cancel 0, Response 0, Prack 0, Comet 0, Reliable1xx 0, Notify 0 SDP application statistics: Parses: 0, Builds 0 Invalid token order: 0, Invalid param: 0 Not SDP desc: 0, No resource: 0

Step 4

show sip-ua status Use this command to display status for the SIP user agent (UA), including whether call redirection is enabled or disabled.
Router# show sip-ua status SIP User Agent Status SIP User Agent for UDP : ENABLED SIP User Agent for TCP : ENABLED SIP User Agent bind status(signaling): DISABLED SIP User Agent bind status(media): DISABLED SIP max-forwards : 6 SIP DNS SRV version: 1 (rfc 2052) Redirection (3xx) message handling: ENABLED

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Step 5

show sip-ua timers Use this command to display the current settings for the SIP user-agent (UA) timers. The following sample output shows the waiting time before a register request is sentthat is, the value that is set with the timers register command:
Router# show sip-ua timers SIP UA Timer Values (millisecs) trying 500, expires 180000, connect 500, disconnect 500 comet 500, prack 500, rel1xx 500, notify 500 refer 500, register 500

General Troubleshooting Tips

Note

For more information on troubleshooting, see the following references:

Cisco IOS Voice Troubleshooting and Monitoring Guide Cisco Technical Support at http://www.cisco.com/en/US/support/index.html Cisco IOS Debug Command Reference, Release 12.3T Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 Troubleshooting and Debugging VoIP Call Basics Voice Gateway Error Decoder for Cisco IOS VoIP Debug Commands

Note

Commands are listed in alphabetical order.

Make sure that VoIP is working. Make sure that you can make a voice call. Verify that SIP-supported codecs are used. Support for codecs varies on different platforms; use the codec ? command to determine the codecs available on a specific platform. Use the debug aaa authentication command to display high-level diagnostics related to AAA logins. Use the debug asnl events command to verify that the SIP subscription server is up. The output displays a pending message if, for example, the client is unsuccessful in communicating with the server. Use the debug call fallback family of commands to display details of VoIP call fallback. Use the debug cch323 family of commands to provide debugging output for various components within an H.323 subsystem. Use the debug ccsip family of commands for general SIP debugging, including viewing direction-attribute settings and port and network address-translation traces. Use any of the following related commands:

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debug ccsip allEnables all SIP-related debugging debug ccsip callsEnables tracing of all SIP service-provider interface (SPI) calls debug ccsip errorEnables tracing of SIP SPI errors debug ccsip eventsEnables tracing of all SIP SPI events debug ccsip infoEnables tracing of general SIP SPI information, including verification that

call redirection is disabled

debug ccsip mediaEnables tracing of SIP media streams debug ccsip messagesEnables all SIP SPI message tracing, such as those that are exchanged

between the SIP user-agent client (UAC) and the access server
debug ccsip preauthEnables diagnostic reporting of authentication, authorization, and

accounting (AAA) preauthentication for SIP calls

debug ccsip statesEnables tracing of all SIP SPI state tracing debug ccsip transportEnables tracing of the SIP transport handler and the TCP or User

Datagram Protocol (UDP) process

Use the debug isdn q931 command to display information about call setup and teardown of ISDN network connections (layer 3) between the local router (user side) and the network. Use the debug kpml command to enable debug tracing of KeyPad Markup Language (KPML) parser and builder errors. Use the debug radius command to enable debug tracing of RADIUS attributes. Use the debug rpms-proc preauth command to enable debug tracing on the RPMS process for H.323 calls, SIP calls, or both H.323 and SIP calls. Use the debug rtr trace command to trace the execution of an SAA operation. Use the debug voip family of commands, including the following:
debug voip ccapi protoheadersDisplays messages sent between the originating and

terminating gateways. If no headers are being received by the terminating gateway, verify that the header-passing command is enabled on the originating gateway.
debug voip ivr scriptDisplays any errors that might occur when the Tcl script is run debug voip rtp session named-event 101Displays information important to DTMF-relay

debugging, if you are using codec types g726r16 or g726r24. Be sure to append the argument 101 to the command to prevent the console screen from flooding with messages and all calls from failing. Sample output for some of these commands follows:

Sample Output for the debug ccsip events Command, page 57 Sample Output for the debug ccsip info Command, page 58

Sample Output for the debug ccsip events Command

The example shows how the Proxy-Authorization header is broken down into a decoded username and password.
Router# debug ccsip events CCSIP SPI: SIP Call Events tracing is enabled 21:03:21: sippmh_parse_proxy_auth: Challenge is 'Basic'. 21:03:21: sippmh_parse_proxy_auth: Base64 user-pass string is 'MTIzNDU2Nzg5MDEyMzQ1Njou'.

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21:03:21: sip_process_proxy_auth: Decoded user-pass string is '1234567890123456:.'. 21:03:21: sip_process_proxy_auth: Username is '1234567890123456'. 21:03:21: sip_process_proxy_auth: Pass is '.'. 21:03:21: sipSPIAddBillingInfoToCcb: sipCallId for billing records = 10872472-173611CC-81E9C73D-F836C2B6@172.18.192.19421:03:21: ****Adding to UAS Request table

Sample Output for the debug ccsip info Command

This example shows only the portion of the debug output that shows that call redirection is disabled. When call redirection is enabled (default), there are no debug line changes.
Router# debug ccsip info 00:20:32: :5060 00:20:32: 00:20:32: 00:20:32: 00:20:32: 00:20:32: 00:20:32: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 172.18.207.10 CCSIP-SPI-CONTROL: act_sentinvite_new_message CCSIP-SPI-CONTROL: sipSPICheckResponse sip_stats_status_code ccsip_get_code_class: !!Call Redirection feature is disabled on the GW ccsip_map_call_redirect_responses: !!Mapping 302 response to 480 Roundtrip delay 4 milliseconds for method INVITE

Configuration Examples for Basic SIP Configuration

This section provides the following configuration examples:

SIP Register Support: Example, page 58 SIP Redirect Processing Enhancement: Examples, page 60 SIP 300 Multiple Choice Messages: Example, page 64

SIP Register Support: Example

Current configuration : 3394 bytes ! version 12.2 service timestamps debug uptime service timestamps log uptime no service password-encryption service internal ! memory-size iomem 15 ip subnet-zero ! no ip domain lookup ! voice service voip redirect ip2ip sip redirect contact order best-match ip dhcp pool vespa network 192.168.0.0 255.255.255.0 option 150 ip 192.168.0.1 default-router 192.168.0.1 ! voice call carrier capacity active !

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voice class codec 1 codec preference 2 g711ulaw ! no voice hpi capture buffer no voice hpi capture destination ! fax interface-type fax-mail mta receive maximum-recipients 0 ! interface Ethernet0/0 ip address 10.8.17.22 255.255.0.0 half-duplex ! interface FastEthernet0/0 ip address 192.168.0.1 255.255.255.0 speed auto no cdp enable h323-gateway voip interface h323-gateway voip id vespa2 ipaddr 10.8.15.4 1718 ! router rip network 10.0.0.0 network 192.168.0.0 ! ip default-gateway 10.8.0.1 ip classless ip route 0.0.0.0 0.0.0.0 10.8.0.1 no ip http server ip pim bidir-enable ! tftp-server flash:SEPDEFAULT.cnf tftp-server flash:P005B302.bin call fallback active ! call application global default.new call rsvp-sync ! voice-port 1/0 ! voice-port 1/1 ! mgcp profile default ! dial-peer voice 1 pots destination-pattern 5100 port 1/0 ! dial-peer voice 2 pots destination-pattern 9998 port 1/1 ! dial-peer voice 123 voip destination-pattern [12]... session protocol sipv2 session target ipv4:10.8.17.42 dtmf-relay sip-notify ! gateway ! sip-ua retry invite 3 retry register 3 timers register 150 registrar dns:myhost3.example.com expires 3600

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registrar ipv4:10.8.17.40 expires 3600 secondary ! telephony-service max-dn 10 max-conferences 4 ! ephone-dn 1 number 4001 ! ephone-dn 2 number 4002 ! line con 0 exec-timeout 0 0 line aux 0 line vty 0 4 login line vty 5 15 login ! no scheduler allocate end

SIP Redirect Processing Enhancement: Examples

This section provides configuration examples to match the identified configuration tasks in the previous sections.

Call Redirection Disabled, page 60 Call Redirection Enabled, page 61 Call Redirection Using IP-to-IP Redirection, page 62 SIP 300 Multiple Choice Messages: Example, page 64

Note

IP addresses and hostnames in examples are fictitious.

Call Redirection Disabled

This example shows that call redirection is disabled on the gateway.

Router# show running-config Building configuration... Current configuration : 2791 bytes ! version 12.2 service config no service single-slot-reload-enable no service pad service timestamps debug uptime service timestamps log uptime no service password-encryption service internal service udp-small-servers ! interface FastEthernet2/0 ip address 172.18.200.24 255.255.255.0 duplex auto

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no shut speed 10 ip rsvp bandwidth 7500 7500 ! voice-port 1/1/1 no supervisory disconnect lcfo ! dial-peer voice 1 pots application session destination-pattern 8183821111 port 1/1/1 ! dial-peer voice 3 voip application session destination-pattern 7173721111 session protocol sipv2 session target ipv4:172.18.200.36 codec g711ulaw ! dial-peer voice 4 voip application session destination-pattern 6163621111 session protocol sipv2 session target ipv4:172.18.200.33 codec g711ulaw ! gateway ! sip-ua no redirection retry invite 1 retry bye 1 ! line con 0 line aux 0 line vty 0 4 login ! end

Call Redirection Enabled

This example shows that call redirection is enabled on the gateway (the default). WHen call redirection is enabled, the output shows no redirection.
Router# show running-config Building configuration... Current configuration : 2791 bytes ! version 12.2 service config no service single-slot-reload-enable no service pad service timestamps debug uptime service timestamps log uptime no service password-encryption service internal service udp-small-servers ! interface FastEthernet2/0 ip address 172.18.200.24 255.255.255.0 duplex auto

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no shut speed 10 ip rsvp bandwidth 7500 7500 ! voice-port 1/1/1 no supervisory disconnect lcfo ! dial-peer voice 1 pots application session destination-pattern 8183821111 port 1/1/1 ! dial-peer voice 3 voip application session destination-pattern 7173721111 session protocol sipv2 session target ipv4:172.18.200.36 codec g711ulaw ! dial-peer voice 4 voip application session destination-pattern 6163621111 session protocol sipv2 session target ipv4:172.18.200.33 codec g711ulaw ! gateway ! sip-ua retry invite 1 retry bye 1 ! line con 0 line aux 0 line vty 0 4 login ! end

Call Redirection Using IP-to-IP Redirection

This example shows that redirection was set globally on the router.
Current configuration : 3394 bytes ! version 12.2 service timestamps debug uptime service timestamps log uptime no service password-encryption service internal ! memory-size iomem 15 ip subnet-zero ! no ip domain lookup ! voice service voip redirect ip2ip sip redirect contact order best-match ip dhcp pool vespa network 192.168.0.0 255.255.255.0 option 150 ip 192.168.0.1

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default-router 192.168.0.1 ! voice call carrier capacity active ! voice class codec 1 codec preference 2 g711ulaw ! ! no voice hpi capture buffer no voice hpi capture destination ! fax interface-type fax-mail mta receive maximum-recipients 0 ! interface Ethernet0/0 ip address 10.8.17.22 255.255.0.0 half-duplex ! interface FastEthernet0/0 ip address 192.168.0.1 255.255.255.0 speed auto no cdp enable h323-gateway voip interface h323-gateway voip id vespa2 ipaddr 10.8.15.4 1718 ! router rip network 10.0.0.0 network 192.168.0.0 ! ip default-gateway 10.8.0.1 ip classless ip route 0.0.0.0 0.0.0.0 10.8.0.1 no ip http server ip pim bidir-enable ! tftp-server flash:SEPDEFAULT.cnf tftp-server flash:P005B302.bin call fallback active ! ! call application global default.new call rsvp-sync ! voice-port 1/0 ! voice-port 1/1 ! mgcp profile default ! dial-peer voice 1 pots destination-pattern 5100 port 1/0 ! dial-peer voice 2 pots destination-pattern 9998 port 1/1 ! dial-peer voice 123 voip destination-pattern [12]... session protocol sipv2 session target ipv4:10.8.17.42 dtmf-relay sip-notify ! gateway

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! sip-ua retry invite 3 retry register 3 timers register 150 registrar dns:myhost3.example.com expires 3600 registrar ipv4:10.8.17.40 expires 3600 secondary ! ! telephony-service max-dn 10 max-conferences 4 ! ephone-dn 1 number 4001 ! ephone-dn 2 number 4002 ! line con 0 exec-timeout 0 0 line aux 0 line vty 0 4 login line vty 5 15 login ! no scheduler allocate end

SIP 300 Multiple Choice Messages: Example

This section provides a configuration example showing redirect contact order set to best match.
Current configuration : 3394 bytes ! version 12.2 service timestamps debug uptime service timestamps log uptime no service password-encryption service internal ! memory-size iomem 15 ip subnet-zero ! no ip domain lookup ! voice service voip redirect ip2ip sip redirect contact order best-match ip dhcp pool vespa network 192.168.0.0 255.255.255.0 option 150 ip 192.168.0.1 default-router 192.168.0.1 ! voice call carrier capacity active ! voice class codec 1 codec preference 2 g711ulaw !

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no voice hpi capture buffer no voice hpi capture destination ! fax interface-type fax-mail mta receive maximum-recipients 0 ! interface Ethernet0/0 ip address 10.8.17.22 255.255.0.0 half-duplex ! interface FastEthernet0/0 ip address 192.168.0.1 255.255.255.0 speed auto no cdp enable h323-gateway voip interface h323-gateway voip id vespa2 ipaddr 10.8.15.4 1718 ! router rip network 10.0.0.0 network 192.168.0.0 ! ip default-gateway 10.8.0.1 ip classless ip route 0.0.0.0 0.0.0.0 10.8.0.1 no ip http server ip pim bidir-enable ! tftp-server flash:SEPDEFAULT.cnf tftp-server flash:P005B302.bin call fallback active ! call application global default.new call rsvp-sync ! voice-port 1/0 ! voice-port 1/1 ! mgcp profile default ! dial-peer voice 1 pots destination-pattern 5100 port 1/0 ! dial-peer voice 2 pots destination-pattern 9998 port 1/1 ! dial-peer voice 123 voip destination-pattern [12]... session protocol sipv2 session target ipv4:10.8.17.42 dtmf-relay sip-notify ! gateway ! sip-ua retry invite 3 retry register 3 timers register 150 registrar dns:myhost3.example.com expires 3600 registrar ipv4:10.8.17.40 expires 3600 secondary ! telephony-service

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max-dn 10 max-conferences 4 ! ephone-dn 1 number 4001 ! ephone-dn 2 number 4002 ! line con 0 exec-timeout 0 0 line aux 0 line vty 0 4 login line vty 5 15 login ! no scheduler allocate end

Additional References

SIP Features Roadmap on page 1Describes how to access Cisco Feature Navigator; also lists and describes, by Cisco IOS release, SIP features for that release.

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