DTMF Events through SIP Signaling
Download this chapter
DTMF Events through SIP SignalingDTMF Events through SIP Signaling
 Feedback

Table Of Contents

DTMF Events through SIP Signaling

Prerequisites

Restrictions

Configuring DTMF Events through SIP Signaling

Troubleshooting Tips


DTMF Events through SIP Signaling

The DTMF Events through SIP Signaling feature provides the following:

•DTMF event notification for SIP messages.

•Capability of receiving hookflash event notification through the SIP NOTIFY method.

•Third-party call control, or other signaling mechanisms, to provide enhanced services, such as calling card and messaging services.

•Communication with the application outside of the media connection.

The DTMF Events through SIP Signaling feature allows telephone event notifications to be sent through SIP NOTIFY messages, using the SIP SUBSCRIBE/NOTIFY method as defined in the Internet Engineering Task Force (IETF) draft, SIP-Specific Event Notification.

The feature also supports sending DTMF notifications based on the IETF draft: Signaled Telephony Events in the Session Initiation Protocol (SIP) (draft-mahy-sip-signaled-digits-01.txt).

Prerequisites

Cisco Unified Border Element

•Cisco IOS Release 12.2(11)T or a later release must be installed and running on your Cisco Unified Border Element.

Cisco Unified Border Element (Enterprise)

•Cisco IOS XE Release 2.5 or a later release must be installed and running on your Cisco ASR 1000 Series Router.

Restrictions

The DTMF Events through SIP Signaling feature adds support for sending telephone-event notifications via SIP NOTIFY messages from a SIP gateway. The events for which notifications are sent out are DTMF events from the local Plain Old Telephone Service (POTS) interface on the gateway. Notifications are not sent for DTMF events received in the Real-Time Transport Protocol (RTP) stream from the recipient user agent.

Configuring DTMF Events through SIP Signaling

To configure the DTMF Events through SIP Signaling feature, perform the following steps.

SUMMARY STEPS

1. enable

2. configure terminal

3. sip-ua

4. timers notify number

5. retry notify number

6. exit

DETAILED STEPS
 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enters privileged EXEC mode or any other security level set by a system administrator.

•Enter your password if prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

sip-ua

Example:

Router(config)# sip-ua

Enters SIP user-agent configuration mode.

Step 4 

timers notify number

Example:

Router(config-sip-ua)# timers notify 100

Sets the amount of time that the user agent waits before retransmitting the Notify message. The argument is as follows:

•number—Time, in milliseconds, to wait before retransmitting. Range: 100 to 1000. Default: 500.

Step 5 

retry notify number

Example:

Router(config-sip-ua)# retry notify 6

Sets the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request. The argument is as follows:

•number—Number of retries. Range: 1 to 10. Default: 10.

Step 6 

exit

Example:

Router(config-sip-ua)# exit

Exits the current mode.

Verifying SIP DTMF Support

To verify SIP DTMF support, perform the following steps as appropriate (commands are listed in alphabetical order).

SUMMARY STEPS

1. show running-config

2. show sip-ua retry

3. show sip-ua statistics

4. show sip-ua status

5. show sip-ua timers

6. show voip rtp connections

7. show sip-ua calls

DETAILED STEPS

Step 1 show running-config

Use this command to show dial-peer configurations.

The following sample output shows that the dtmf-relay sip-notify command is configured in dial peer 123: 

Router# show running-config


.

.

.

dial-peer voice 123 voip

 destination-pattern [12]...

 monitor probe icmp-ping

 session protocol sipv2

 session target ipv4:10.8.17.42

 dtmf-relay sip-notify

The following sample output shows that DTMF relay and NTE are configured on the dial peer. 

Router# show running-config


!

dial-peer voice 1000 pots

 destination-pattern 4961234

 port 1/0/0

!

dial-peer voice 2000 voip

 application session

 destination-pattern 4965678

 session protocol sipv2

 session target ipv4:192.0.2.34

 dtmf-relay rtp-nte

! RTP payload type value = 101 (default)

!

dial-peer voice 3000 voip

 application session

 destination-pattern 2021010101

 session protocol sipv2

 session target ipv4:192.0.2.34

 dtmf-relay rtp-nte

 rtp payload-type nte 110

! RTP payload type value = 110 (user assigned)

!

Step 2 show sip-ua retry

Use this command to display SIP retry statistics. 

Router# show sip-ua retry


SIP UA Retry Values

invite retry count = 6 response retry count = 1

bye retry count = 1 cancel retry count = 1

prack retry count = 10 comet retry count = 10

reliable 1xx count = 6 notify retry count = 10

Step 3 show sip-ua statistics

Use this command to display response, traffic, and retry SIP statistics.

Tip To reset counters for the show sip-ua statistics display, use the clear sip-ua statistics command. 

Router# show sip-ua statistics


SIP Response Statistics (Inbound/Outbound)

Informational:

Trying 4/2, Ringing 2/1,

Forwarded 0/0, Queued 0/0,

SessionProgress 0/0

Success:

OkInvite 1/2, OkBye 0/1,

OkCancel 1/0, OkOptions 0/0,

OkPrack 2/0, OkPreconditionMet 0/0,

OkNotify 1/0, 202Accepted 0/1

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

RequestCancel 1/0, NotAcceptableMedia 0/0

Server Error:

InternalError 0/1, 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

SIP Total Traffic Statistics (Inbound/Outbound) /* Traffic Statistics

Invite 3/2, Ack 3/2, Bye 1/0,

Cancel 0/1, Options 0/0,

Prack 0/2, Comet 0/0,

Notify 0/1, Refer 1/0

Retry Statistics 							/* Retry Statistics

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

Prack 0, Comet 0, Reliable1xx 0, Notify 0

Following is sample output verifying configuration of the SIP INFO Method for DTMF Tone Generation feature: 

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

Step 4 show sip-ua status

Use this command to display status for the SIP user agent. 

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

The following sample output shows that the time interval between consecutive NOTIFY messages for a telephone event is the default of 2000 ms: 

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 early-media for 180 responses with SDP: ENABLED

SIP max-forwards : 6

SIP DNS SRV version: 2 (rfc 2782)

NAT Settings for the SIP-UA

Role in SDP: NONE

Check media source packets: DISABLED

Maximum duration for a telephone-event in NOTIFYs: 2000 ms

SIP support for ISDN SUSPEND/RESUME: ENABLED

Redirection (3xx) message handling: ENABLED


 SDP application configuration:

 Version line (v=) required

 Owner line (o=) required

 Timespec line (t=) required

 Media supported: audio image

 Network types supported: IN

 Address types supported: IP4

 Transport types supported: RTP/AVP udptl

The following sample output shows configuration of the SIP INFO Method for DTMF Tone Generation feature: 

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

Step 5 show sip-ua timers

Use this command to display the current settings for SIP user-agent timers. 

Router# show sip-ua timers


SIP UA Timer Values (millisecs)

trying 500, expires 300000, connect 500, disconnect 500

comet 500, prack 500, rel1xx 500, notify 500

Step 6 show voip rtp connections

Use this command to show local and remote Calling ID and IP address and port information.

Step 7 show sip-ua calls

Use this command to ensure the DTMF method is SIP-KPML.

The following sample output shows that the DTMF method is SIP-KPML. 

router# show sip-ua calls


SIP UAC CALL INFO


Call 1

SIP Call ID                : 57633F68-2BE011D6-8013D46B-B4F9B5F6@172.18.193.251

   State of the call       : STATE_ACTIVE (7)

   Substate of the call    : SUBSTATE_NONE (0)

   Calling Number          : 

   Called Number           : 8888

   Bit Flags               : 0xD44018 0x100 0x0

   CC Call ID              : 6

   Source IP Address (Sig ): 192.0.2.1

   Destn SIP Req Addr:Port : 192.0.2.2:5060

   Destn SIP Resp Addr:Port: 192.0.2.3:5060

   Destination Name        : 192.0.2.4.250

   Number of Media Streams : 1

   Number of Active Streams: 1

   RTP Fork Object         : 0x0

   Media Mode              : flow-through

   Media Stream 1

     State of the stream      : STREAM_ACTIVE

     Stream Call ID           : 6

     Stream Type              : voice-only (0)

     Negotiated Codec         : g711ulaw (160 bytes)

	Codec Payload Type       : 0 

     Negotiated Dtmf-relay    : sip-kpml

     Dtmf-relay Payload Type  : 0

     Media Source IP Addr:Port: 192.0.2.5:17576

     Media Dest IP Addr:Port  : 192.0.2.6:17468

     Orig Media Dest IP Addr:Port : 0.0.0.0:0


   Number of SIP User Agent Client(UAC) calls: 1


SIP UAS CALL INFO


   Number of SIP User Agent Server(UAS) calls: 0

Troubleshooting Tips

•To enable debugging for RTP named-event packets, use the debug voip rtp command.

•To enable KPML debugs, use the debug kpml command.

•To enable SIP debugs, use the debug ccsip command.

•Collect debugs while the call is being established and during digit presses.

•If an established call is not sending digits through KPML, use the show sip-ua calls command to ensure SIP-KPML is included in the negotiation process.