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Cisco IOS Software Releases 12.2 T

SIP INFO Method for DTMF Tone Generation

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Table Of Contents

SIP INFO Method for DTMF Tone Generation

Feature Overview

SIP INFO Messages

Benefits

Restrictions

Related Features and Technologies

Related Documents

Supported Platforms

Supported Standards, MIBs, and RFCs

Prerequisites

Configuration Tasks

Verifying the SIP INFO Method for DTMF Tone Generation Feature

Command Reference

show sip-ua

Glossary


SIP INFO Method for DTMF Tone Generation

Document Update Alert

This document was originally produced for Cisco IOS Release 12.2(11)T. This feature has been updated in subsequent releases, and more recent documentation is available.

If you are using Cisco IOS Release 12.2(11)T or higher, refer to the following section in the Configuring Additional SIP Features chapter of the Cisco IOS SIP Configuration Guide, Cisco IOS Voice Configuration Library, Release 12.3:

•SIP INFO Method for DTMF Tone Generation

Feature History

Release
Modification

12.2(11)T

This feature was introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms.


This document describes the SIP INFO Method for DTMF Tone Generation feature in Cisco IOS Release 12.2(11)T, including information on the benefits of the new feature, supported platforms, and related documents.

This document includes the following sections:

•Feature Overview

•Supported Platforms

•Supported Standards, MIBs, and RFCs

•Prerequisites

•Configuration Tasks

•Command Reference

•Glossary

Feature Overview

This document describes the SIP INFO Method for DTMF Tone Generation feature, which uses the Session Initiation Protocol (SIP) INFO method to generate dual tone multi frequency (DTMF) tones on the telephony call leg. SIP methods, or request message types, request a specific action be taken by another user agent (UA) or proxy server. The SIP INFO message is sent along the signaling path of the call. With the new feature, upon receipt of a SIP INFO message with DTMF relay content, the gateway generates the specified DTMF tone on the telephony end of the call.

The SIP INFO Method for DTMF Tone Generation feature is always enabled, and is invoked when a SIP INFO message is received with DTMF relay content. This feature is related to the DTMF Events Through SIP Signaling feature, which provides the ability for an application to be notified about DTMF events using SIP NOTIFY messages. Together, the two features provide a mechanism to both send and receive DTMF digits along the signaling path. For more information on sending DTMF event notification using SIP NOTIFY messages, refer to the DTMF Events Through SIP Signaling feature.

SIP INFO Messages

The SIP INFO method is used by a UA to send call signaling information to another UA with which it has an established media session. The following example shows a SIP INFO message with DTMF content:

INFO sip:2143302100@172.17.2.33 SIP/2.0
Via: SIP/2.0/UDP 172.80.2.100:5060
From: <sip:9724401003@172.80.2.100>;tag=43
To: <sip:2143302100@172.17.2.33>;tag=9753.0207
Call-ID: 984072_15401962@172.80.2.100
CSeq: 25634 INFO
Supported: 100rel
Supported: timer
Content-Length: 26
Content-Type: application/dtmf-relay
Signal= 1
Duration= 160

This sample message shows a SIP INFO message received by the gateway with specifics about the DTMF tone to be generated. The combination of the From, To, and Call-ID headers identifies the call leg. The signal and duration headers specify the digit, in this case 1, and duration, 160 milliseconds in the example, for DTMF tone play.

Benefits

The SIP INFO Method for DTMF Tone Generation feature provides the following benefits:

•Provides DTMF tone generation for SIP requests

•Enables transport of DTMF digits along the signaling path

Restrictions

The SIP INFO Method for DTMF Tone Generation feature includes the following signal duration parameters:

•Minimum signal duration is 100 milliseconds (ms). If a request is received with a duration less than 100ms, the minimum duration of 100 ms is used by default.

•Maximum signal duration is 5000 ms. If a request is received with a duration longer than 5000 ms, the maximum duration of 5000 ms is used by default.

•If no duration parameter is included in a request, the gateway defaults to a signal duration of 250 ms.

Related Features and Technologies

•DTMF Events Through SIP Signaling feature

•Cisco SIP Proxy Server

•Cisco Voice over IP (VoIP)

Related Documents

•RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

•RFC 2543, SIP: Session Initiation Protocol

•RFC 2976, SIP INFO Method

•Session Initiation Protocol Gateway Call Flows and Compliance Information

•Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2

•Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2

•Cisco SIP Proxy Server Administrator Guide, Version 1.2, "Configuring the Cisco SIP Proxy Server" chapter

Supported Platforms

•Cisco 2600 series

•Cisco 3600 series

•Cisco AS5300

•Cisco AS5350

•Cisco AS5400

•Cisco AS5850

•Cisco 7200 series

Determining Platform Support Through Cisco Feature Navigator

Cisco IOS software is packaged in feature sets that are supported on specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.

Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.

To access Cisco  Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:

http://www.cisco.com/register

Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:

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

Availability of Cisco IOS Software Images

Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.

Supported Standards, MIBs, and RFCs

Standards

No new or modified standards are supported by this feature.

MIBs

No new or modified MIBS are supported by this feature.

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

http://tools.cisco.com/ITDIT/MIBS/servlet/index

If Cisco  MIB Locator does not support the MIB information that you need, you can also obtain a list of supported MIBs and download MIBs from the Cisco  MIBs page at the following URL:

http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml

To access Cisco MIB Locator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:

http://www.cisco.com/register

RFCs

•RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

•RFC 2543, SIP: Session Initiation Protocol

•RFC 2976, SIP INFO Method

Prerequisites

The following are general prerequisites for SIP functionality:

•Ensure that your Cisco 2600 series, Cisco 3600 series, or Cisco 7200 series router has at least 16 MB Flash memory and 64 MB DRAM. A Cisco AS5300 must have a minimum of 16 MB Flash memory and 128 MB DRAM. A Cisco AS5400 must have a minimum of 32 MB Flash memory and 256 MB DRAM.

•Ensure that the gateway has voice functionality that is configured for SIP.

For more information about configuring SIP, refer to

Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2, the "Configuring Session Initiation Protocol for Voice over IP" chapter.

•Establish a working IP network.

For more information about configuring IP, refer to

Cisco IOS IP Configuration Guide, Release 12.2.

•Configure VoIP.

For more information about configuring VoIP, refer to

Session Initiation Protocol for VoIP on Cisco Access Platforms.

Configuration Tasks

None

Verifying the SIP INFO Method for DTMF Tone Generation Feature

You cannot configure, enable, or disable this feature. You can display SIP statistics, including SIP INFO method statistics, by using the show sip-ua statistics and show sip-ua status commands in privileged exec mode. See the following fields for SIP INFO method statistics:

•OkInfo 0/0, under SIP Response Statistics, Success, displays the number of successful responses to an INFO request.

•Info 0/0, under SIP Total Traffic Statistics, displays the number of INFO messages received and sent by the gateway.

The following is sample output from the show sip-ua statistics command: 

router# show sip-ua statistics


SIP Response Statistics (Inbound/Outbound)

Informational:

Trying 1/1, Ringing 0/0,

Forwarded 0/0, Queued 0/0,

SessionProgress 0/1

Success:

OkInvite 0/1, OkBye 1/0,

OkCancel 0/0, OkOptions 0/0,

OkPrack 0/0, OkPreconditionMet 0/0

OkSubscibe 0/0, OkNotify 0/0,

OkInfo 0/0, 202Accepted 0/0

Redirection (Inbound only):

MultipleChoice 0, MovedPermanently 0,

MovedTemporarily 0, SeeOther 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,

LengthRequired 0/0, ReqEntityTooLarge 0/0,

ReqURITooLarge 0/0, UnsupportedMediaType 0/0,

BadExtension 0/0, TempNotAvailable 0/0,

CallLegNonExistent 0/0, LoopDetected 0/0,

TooManyHops 0/0, AddrIncomplete 0/0,

Ambiguous 0/0, BusyHere 0/0,

BadEvent 0/0

Server Error:

InternalError 0/0, NotImplemented 0/0,

BadGateway 0/0, ServiceUnavail 0/0,

GatewayTimeout 0/0, BadSipVer 0/0

Global Failure:

BusyEverywhere 0/0, Decline 0/0,

NotExistAnywhere 0/0, NotAcceptable 0/0

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

Retry Statistics

Invite 0, Bye 0, Cancel 0, Response 0, Notify 0

The following is sample output from 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 

SIP User Agent bind status(signaling): DISABLED 

SIP User Agent bind status(media): DISABLED 

SIP max-forwards : 6 

SIP DNS SRV version: 2 (rfc 2782) 

SDP application configuration: 

 Version line (v=) required 

 Owner line (o=) required 

 Session name line (s=) required 

 Timespec line (t=) required 

 Media supported: audio image 

 Network types supported: IN 

 Address types supported: IP4 

 Transport types supported: RTP/AVP udptl

Note In addition to the supported platforms listed elsewhere in this document, the SDP application configuration output also displays on the Cisco 2400 series and the Cisco MC3810 platforms.

Command Reference

This section documents modified commands associated with this feature. All other commands used with this feature are documented in the Cisco IOS Release 12.2 command reference publications.

•show sip-ua

show sip-ua

To display information and settings for the Session Initiation Protocol (SIP) User Agent (UA), use the show sip-ua command in privileged EXEC mode.

show sip-ua {retry | statistics | status | timers}

Syntax Description

retry

Displays SIP protocol retry counts.

statistics

Displays SIP UA response, traffic, and retry statistics.

status

Displays SIP UA listener status.

timers

Displays current settings for the SIP UA protocol timers.


Defaults

There are no default behaviors or values for this command.

Command Modes

Privileged EXEC

Command History

Release
Modification

12.1(1)T

This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server

12.1(3)T

The following changes were made:

•The statistics keyword was added.

•The statistics portion of the output from the status keyword was moved from the status keyword to the statistics keyword

•The output from the timers keyword was changed to reflect the changes in the timers command.

12.2(2)XB

The output from the statistics keyword was enhanced to display:

•BadRequest counter (400 class) now counts Malformed Via entries.

•Reliable provisional responses (PRACK/rel1xx).

•Conditions met (COMET).

•NOTIFY responses.

12.2.(11)T

The output for the statistics keyword was enhanced to display:

•OkInfo counter (200) class now counts the number of successful responses to INFO requests.

•Info counter now counts the number of INFO messages received and sent.

•BadEvent counter (489 response) now counts responses to SUBSCRIBE requests with event types that are not understood by the server.

•OkSubscribe counter (200 class) now counts the number of 200 OK SIP messages received and sent in response to Subscribe messages.

•Subscribe requests now indicates total requests received and sent.

•SDP application statistics.

The output for the status keyword was enhanced to display SDP application configuration. In addition to the supported platforms listed elsewhere in this document, the SDP application configuration and SDP statistics output is displayed on the Cisco 2400 series and the Cisco MC3810 platforms.


Usage Guidelines

Use the show sip-ua command to verify SIP configurations.

Examples

The following is sample output from the show sip-ua statistics command: 

router# show sip-ua statistics


SIP Response Statistics (Inbound/Outbound)

Informational:

Trying 1/1, Ringing 0/0,

Forwarded 0/0, Queued 0/0,

SessionProgress 0/1

Success:

OkInvite 0/1, OkBye 1/0,

OkCancel 0/0, OkOptions 0/0,

OkSubscibe 0/0, OkNotify 0/0,

OkInfo 0/0, 202Accepted 0/0

Redirection (Inbound only):

MultipleChoice 0, MovedPermanently 0,

MovedTemporarily 0, SeeOther 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,

LengthRequired 0/0, ReqEntityTooLarge 0/0,

ReqURITooLarge 0/0, UnsupportedMediaType 0/0,

BadExtension 0/0, TempNotAvailable 0/0,

CallLegNonExistent 0/0, LoopDetected 0/0,

TooManyHops 0/0, AddrIncomplete 0/0,

Ambiguous 0/0, BusyHere 0/0,

BadEvent 0/0

Server Error:

InternalError 0/0, NotImplemented 0/0,

BadGateway 0/0, ServiceUnavail 0/0,

GatewayTimeout 0/0, BadSipVer 0/0

Global Failure:

BusyEverywhere 0/0, Decline 0/0,

NotExistAnywhere 0/0, NotAcceptable 0/0


SIP Total Traffic Statistics (Inbound/Outbound)

Invite 1/0, Ack 1/0, Bye 0/1,

Cancel 0/0, Options 0/0, Subscribe 0/0, Notify 0/0

Refer 0/0, Info 0/0


Retry Statistics

Invite 0, Bye 0, Cancel 0, Response 0, Notify 0


SDP application statistics: 

Parses:4, Builds 2 

Invalid token order:0, Invalid param:0 

Not SDP desc:0, No resource:0

The show sip-ua statistics command generates output that includes a reason phrase and a count describing the SIP messages received and sent. When 0/0 is included in a field, the first number is an inbound count, and the second number is an outbound count. The description field, shown in Table 1which follows, also includes a SIP response code xxx, which the SIP protocol uses in determining behavior. SIP response codes, shown in Table 1 are classified into one of the following six categories:

•1xx: Informational, indicates call progress

•2xx: Success, indicates successful receipt or completion of a request

•3xx: Redirection, a redirect server has returned possible locations

•4xx: Client error, indicates a request cannot be fulfilled as it was submitted

•5xx: Server error, indicates a request has failed due to an error by the server. It may be retried at another server.

•6xx: Global failure, indicates a request has failed and should not be tried again at any server.

Table 1 describes the reason phrase fields and SIP response code descriptions displayed by the show sip-ua statistics command.

Table 1 SIP Response Code Descriptions for the show sip-ua statistics Command

Reason Phrase and Count
SIP Response Code and Description

Accepted 0/0

202 Indicates a successful response to a REFER request received/sent.

Ack 0/0

Indicates a confirmed final response received or sent.

AddrIncomplete 0/0

484 Address supplied is incomplete.

AlternateService 0

380 Unsuccessful call; however, an alternate service is available.

Ambiguous 0/0

485 Address supplied is ambiguous.

BadEvent 0/0

489 Bad Event response indicates a SUBSCRIBE request having an event type that the server could not understand.

BadExtension 0/0

420 Server could not understand the protocol extension in the Require header.

BadGateway 0/0

502 Network is out of order.

BadRequest

400 Bad Request (includes the malformed Via header).

BadSipVer 0/0

505 SIP version requested is not supported.

BusyEverywhere 0/0

600 Called party is busy.

BusyHere 0/0

486 Called party is busy.

Bye 0

Indicates the number of times a Bye request is resent to the other user agent.

Bye 0/0

Terminates the session.

CallLegNonExistent 0/0

481 Server is ignoring the request. It was either a Bye request and there was no matching leg ID, or a Cancel request with no matching transaction.

Cancel 0

Indicates the number of times a Cancel request is resent to the other user agent.

Cancel 0/0

Terminates the pending request.

Comet 0

Indicates the number of times a COMET request is resent to the other user agent.

Comet 0/0

Conditions have been met.

Conflict 0/0

409 Temporary failure.

Decline 0/0

603 Call rejected.

Forbidden 0/0

403 The SIP server has the request, but cannot provide service.

Forwarded 0/0

181 Call has been forwarded.

GatewayTimeout 0/0

504 Indicates that the server or gateway did not receive a timely response from another server (such as a location server).

Gone 0/0

410 Resource is no longer available at the server and no forwarding address is known.

Info 0/0

Indicates the number of times an INFO request is sent and received.

InternalError 0/0

500 Indicates that the server or gateway encountered an unexpected error that prevented it from processing the request.

Invite 0

Indicates the number of times an INVITE request is resent to the other user agent.

Invite 0/0

Initiates a call.

LengthRequired 0/0

411 A content length is required.

LoopDetected 0/0

482 Indicates a loop—server received a request that included itself in the path.

MethodNotAllowed 0/0

405 Method specified in the request is not allowed.

MovedPermanently 0

301 User is no longer available at this location.

MovedTemporarily 0

302 User is temporarily unavailable.

MultipleChoice 0

300 Address resolves to more than one location.

NotAcceptable 0/0

406/606 Call contacted, but some aspect of the session description was unacceptable.

NotAcceptableMedia 0/0

406 Call contacted, but some aspect of the session description was unacceptable.

NotExistAnywhere 0/0

604 Server has authoritative information that the called party does not exist in the network.

NotFound 0/0

404 Called party does not exist in the specified domain.

Notify 0

Indicates the number of times a NOTIFY is resent

to the other user agent.

Notify 0/0

Indicates the number of NOTIFY messages received and sent.

NotImplemented 0/0

501 Service or option not implemented in the server or gateway.

OkBye 0/0

200 Indicates a successful response to a BYE request.

OkCancel 0/0

200 Indicates a successful response to a CANCEL request.

OkInfo

200 A successful response to an INFO request.

OkInvite 0/0

200 Indicates a successful response to an INVITE request.

OkNotify 0/0

200 Indicates a successful response to a NOTIFY request.

OkOptions 0/0

200 Indicates a successful response to an OPTIONS request.

OkPrack 0/0

200 Indicates a successful response to a PRACK request.

OkPreconditionMet 0/0

200 Indicates a successful response to a PRECONDITIONMET request.

OkSubscribe 0/0

200 Indicates a successful response to a SUBSCRIBE request

Options 0/0

Query the receiving or sending server as to its capabilities.

PaymentRequired 0/0

402 Payment is required to complete the call.

Prack 0

Indicates the number of times a PRACK request is resent to the other user agent.

Prack 0/0

Provisional response received or sent.

PreCondFailure 0/0

580 Indicates the session could not be established due to failure to meet required preconditions.

ProxyAuthReqd 0/0

407 Rejected for proxy authentication.

Queued 0/0

182 Until the called party is available, the message is queued.

Refer 0/0

Indicates the number of Refer requests received/sent.

Reliable1xx 0

Indicates the number of times the Reliable 1xx response is resent to the other user agent.

ReqEntityTooLarge 0/0

413 Server refuses to process because the request is larger than is acceptable.

ReqTimeout 0/0

408 Server could not produce a response before the Expires time-out.

RequestCancel 0/0

Request has been canceled.

ReqURITooLarge 0/0

414 Server refuses to process, because the URI (URL) request is too long and cannot be processed correctly.

Response 0

Indicates the number of Response retries.

Retry Statistics

One of the three categories of response statistics.

Ringing 0/0

180 Called party has been located and is being notified of the call.

SeeOther 0

303 Refer to response from the redirect server.

ServiceUnavail 0/0

503 Service option is not available due to an overload or maintenance problem.

SessionProgress 0/0

183 Indicates inband alerting.

SIP Response Statistics (Inbound/Outbound)

One of the three categories of response statistics.

SIP Total Traffic Statistics (Inbound/Outbound)

One of the three categories of response statistics.

Subscribe 0/0

Indicates the number of Subscribe requests received or sent

TempNotAvailable 0/0

480 Called party did not respond.

TooManyHops 0/0

483 Indicates that a server received a request that required more hops than is allowed by the Max-Forward header.

Trying 0/0

100 Action is being taken with no resolution.

UseProxy 0

305 Caller must use a proxy to contact called party.

Unauthorized 0/0

401 Indicates that the request requires user authentication.

UnsupportedMediaType 0/0

415 Server refuses to process because the service option is not available on the destination endpoint.

UseProxy 0

305 Caller must use a proxy to contact the called party.


The following is sample output from 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 

SIP User Agent bind status(signaling): DISABLED 

SIP User Agent bind status(media): DISABLED 

SIP max-forwards : 6 

SIP DNS SRV version: 2 (rfc 2782) 

SDP application configuration: 

 Version line (v=) required 

 Owner line (o=) required 

 Session name line (s=) required 

 Timespec line (t=) required 

 Media supported: audio image 

 Network types supported: IN 

 Address types supported: IP4 

 Transport types supported: RTP/AVP udptl

Related Commands

Command
Description

debug ccsip messages

Displays all SIP SPI message tracing and traces the SIP messages exchanged between the SIP UAC and the access server.


Glossary

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

gateway—A gateway allows SIP or H.323 terminals to communicate with terminals configured to run other protocols. A gateway is the point where a circuit-switched call is encoded and repackaged into IP packets.

originator—User agent that initiates the transfer or Refer request with the recipient.

recipient—User agent that receives the Refer request from the originator and is transferred to the final recipient.

RTP—Real-Time Transport Protocol (RFC 1889)

SIP—Session Initiation Protocol. An application-layer protocol originally developed by the Multiparty Multimedia Session Control (MMUSIC) working group of the Internet Engineering Task Force (IETF). Their goal was to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP features are compliant with IETF RFC 2543, published in March 1999.

UA—user agent.

UAC—user agent client. A client application that initiates a SIP request.

UAS—user agent server (or user agent). 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.