Specifications

Appendix A: Preparing Cisco Unified SRST Support for SIP
DTMF Relay for SIP Applications and Voice Mail
182
Cisco Unified Survivable Remote Site Telephony Version 4.0 System Administrator Guide
DTMF Relay Using SIP RFC 2833
Cisco Skinny Client Control Protocol (SCCP) phones, such as those used with Cisco Unified SRST
systems, provide only out-of-band DTMF digit indications. To enable SCCP phones to send digit
information to remote SIP-based IVR and voice-mail applications, Cisco SRST 3.2 and later versions
provide conversion from the out-of-band SCCP digit indication to the SIP standard for DTMF relay,
which is RFC 2833. You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte
command.
The SIP DTMF relay method is needed in the following situations:
• When SIP is used to connect a Cisco Unified SRST system to a remote SIP-based IVR or voice-mail
application, such as Cisco Unity.
• When SIP is used to connect a Cisco Unified SRST system to a remote SIP-PSTN voice gateway
that goes through the PSTN to a voice-mail or IVR application.
Note The need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones natively
support in-band DTMF relay as specified in RFC 2833.
To enable SIP DTMF relay using RFC 2833, the commands in this section must be used on both
originating and terminating gateways.
SUMMARY STEPS
1. dial-peer voice tag voip
2. dtmf-relay rtp-nte
3. exit
4. sip-ua
5. notify telephone-event max-duration time
6. exit
DETAILED STEPS
Command or Action Purpose
Step 1
dial-peer voice tag voip
Example:
Router(config)# dial-peer voice 2 voip
Enters dial-peer configuration mode.
Step 2
dtmf-relay rtp-nte
Example:
Router(config-dial-peer)# dtmf-relay rtp-nte
Forwards DTMF tones by using Real-Time Transport
Protocol (RTP) with the Named Telephone Event
(NTE) payload type.