DTMF methods and interoperability failures
Verified 2026-09-24 · 57 sources · tier 1–3 · 2 disputed
Dual-tone multi-frequency (DTMF) signaling conveys telephony digits across networks using RTP events, in-band audio tones, or out-of-band SIP messages 5410. Mismatches between endpoints in negotiated transport modes, payload types, or relay configurations lead to interoperability failures such as dropped calls, undetected digits, or duplicated digits 738.
RFC 4733 and RTP event signaling
RFC 4733 (December 2006) on the Standards Track, obsoletes RFC 2833 46. RFC 2833 (May 2000, Proposed Standard) is obsoleted by 2 documents: RFC 4733 and RFC 4734 44. The RFC 4733 payload format for named telephone events is designated 'telephone-event' with media type audio/telephone-event 54.
The telephone-event payload format has no static RTP payload type number; the payload type is assigned dynamically and signalled out-of-band (for example in SDP) 22. WarmTransfer's reading of the sources is that payload type 101 for telephone-event is a common vendor default rather than a value fixed by the standard, so a far end may legitimately negotiate a different dynamic value 42. The default RTP clock rate for the telephone-event payload is 8000 Hz, which can be redefined when the dynamic payload type is assigned 20.
RFC 4733 maps DTMF digits 0-9 to event codes 0-9, star to 10, pound to 11, and A-D to 12-15 21. If no 'events' fmtp parameter is received, an RFC 4733 sender SHOULD assume the peer supports DTMF events 0-15 and no other events 24.
Transmission rules under RFC 4733 require:
- The first RTP packet for a telephone event MUST have the marker (M) bit set 25.
- RFC 4733 allows a source to choose its own spacing for event update packets and recommends 50 ms 55.
- The final (E-bit) packet for each telephone event and each segment SHOULD be sent a total of 3 times at the source's update interval 23.
- If an event lasts beyond the maximum duration field value 0xFFFF, the sender MUST send a packet reporting that maximum without the E bit and continue the event in a new segment 35.
The RFC 4733 volume field expresses power level for tone events from 0 to -63 dBm0 56. RFC 4733 recommends that a gateway render only the telephone-event payload once it is received, because the concurrent audio may contain spurious tones introduced by compression 26.
Sources disagree on the classification of RFC 2833 and RTP event streams: Cisco documentation classifies RFC 2833 NTE as an in-band DTMF method because the events travel in the RTP stream, whereas Asterisk documentation describes rfc4733 DTMF as sent out of band of the main audio stream, and Genesys describes RTP Events as out-of-band 3637.
SIP-level DTMF methods
RFC 6086 (January 2011) defines the SIP INFO method and Info Package framework and obsoletes RFC 2976 47. RFC 6086 states that DTMF transport over SIP INFO has been done with proprietary, unstandardised mechanisms 48. RFC 6086 attributes industry interoperability problems to legacy INFO usages that require static configuration 34. Under RFC 6086 a UA supporting Info Packages MUST list the packages it will accept in the Recv-Info header field 43. RFC 6086 keeps a 'legacy' mode of INFO usage compatible with RFC 2976 for backward compatibility 33.
RFC 4730 (November 2006) defines the 'kpml' SIP event package, in which a subscriber sends digit patterns in a SUBSCRIBE and matched key presses are reported in NOTIFY messages 31.
In-band audio tones and codec limits
In-band DTMF is recognised reliably only on G.711 (ulaw or alaw); low-bit-rate codecs distort the tones so they may not be recognised 30.
Platform implementations and relay behaviors
Platform support for DTMF relay methods and behaviors varies across implementations 102.
Cisco CUBE and Unified CM
On CUBE (IOS XE 17.6 onwards), SIP DTMF relay methods are rtp-nte (RFC 2833), sip-notify (Cisco proprietary), sip-kpml (RFC 4730) and sip-info, plus G.711 in-band audio, which needs no configuration 10. Cisco's proprietary cisco-rtp DTMF relay (payload type 121) has been discontinued on CUBE 5. H.323 (and with it the H.245 alphanumeric and signal DTMF methods) is not supported on CUBE from IOS XE Bengaluru 17.6.1a onwards 9.
On CUBE, DTMF relay is disabled by default on SIP and H.323 dial-peers except SIP INFO, so a relay method has to be configured 13. The CUBE dtmf-relay command accepts up to 6 methods on a VoIP dial-peer 17. When more than one out-of-band DTMF method is configured on a CUBE dial-peer, preference runs from highest to lowest in the order the methods were configured 12. On CUBE, sip-info DTMF handling is always enabled for SIP dial-peers and is invoked when a SIP INFO with DTMF relay content is received 15.
A Cisco IOS XE gateway can receive SIP INFO with Content-Type application/dtmf-relay carrying Signal and Duration fields and play the DTMF tone on the telephony call leg 16. For SIP INFO tone generation the Cisco gateway clamps Duration to 100-5000 ms and uses 250 ms if no duration is given 49.
Interworking failures can occur during relay:
- Digit duplication: When CUBE interworks rtp-nte to an out-of-band method without digit-drop, the same digit is sent both in RTP and out-of-band and the far end sees duplicate digits 7. Cisco's fix for duplicate digits is to configure dtmf-relay rtp-nte digit-drop on the inbound dial-peer and the out-of-band method on the outbound dial-peer, so CUBE suppresses NTE packets and relays only the out-of-band digits 6.
- MTP requirements: If an endpoint supports none of the DTMF relay methods configured on CUBE, an MTP or transcoder is required 11. Cisco's CUBE DTMF technote (updated May 2023) states that Unified CM does not support SIP INFO for DTMF 19. Unified CM inserts an MTP when a SIP endpoint supports only RFC 2833 and the other side needs out-of-band DTMF; the MTP exchanges RFC 2833 packets with the SIP endpoint and out-of-band DTMF with Unified CM 18.
- Webex Calling Local Gateway: Cisco's Webex Calling Local Gateway configuration uses dtmf-relay rtp-nte on the PSTN-facing dial-peer and on the Unified CM-facing dial-peers 57.
Microsoft Teams Phone Direct Routing
The Teams Phone Direct Routing media stack does not support in-band DTMF 51. WarmTransfer's reading of the sources is that the Direct Routing 'RFC standards' list that SBCs must comply with does not name RFC 2833 or RFC 4733, so the page does not itself specify which out-of-band DTMF method or payload type Teams expects 52. WarmTransfer's reading of the sources is that when a PSTN leg delivers DTMF only as in-band audio, the Direct Routing SBC has to convert it to a non-in-band method before it reaches Teams, or Teams IVRs and voicemail will not see the digits 53. For media bypass early media, Microsoft says the SBC should support sending DTMF during the early-media phase to enable IVR and voicemail scenarios 50.
Genesys Cloud
Genesys Cloud external (BYOC) trunks offer 3 DTMF methods: RTP Events (default, RFC 2833/4733), In-band Audio, and None 28. The Genesys Cloud external trunk DTMF Payload setting accepts 96-127, defaults to 101, and applies only when the DTMF Method is RTP Events 29. Genesys lists out-of-band DTMF (RFC 2833) as a required capability for BYOC SIP providers, and says customer endpoints should handle in-band if RFC 2833 is not supported by the PBX or far end 27.
Asterisk and PJSIP
Asterisk res_pjsip endpoint dtmf_mode defaults to rfc4733 and accepts rfc4733, inband, none, info, auto and auto_info 2. In Asterisk, dtmf_mode auto sends RFC 4733 if the other side supports it and in-band otherwise; auto_info sends RFC 4733 if supported and SIP INFO otherwise 1.
PJSIP documentation calls RFC 4733 telephone-event the recommended, standards-based DTMF method 41. In PJSIP, sending RFC 4733 DTMF when the negotiated SDP lacks the telephone-event rtpmap returns PJMEDIA_RTP_EREMNORFC2833; any fallback is up to the application 40. PJSIP generates in-band DTMF tones but does not implement in-band DTMF detection 39. PJSIP documentation warns that if the remote side does not support SIP INFO and never replies, the sender treats it as a transaction timeout and disconnects the call 38.
See also
- See also SDP offer answer and codec negotiation.
- See also Early media and ringback behavior.
- See also SIP transactions dialogs and call setup.
Applicability
Applies to: IETF RTP telephone-event payload, Cisco Unified Border Element, PJSIP project, IETF RFC 2833, RTP telephone-event payload, Sangoma Asterisk res_pjsip, IETF SIP INFO, IETF SIP KPML, in-band audio DTMF, Cisco IOS XE voice gateway, Cisco Unified Communications Manager, Cisco Webex Calling Local Gateway (IOS XE), Microsoft Teams Phone Direct Routing, Microsoft Teams Phone Direct Routing (media bypass), Genesys Cloud BYOC external trunk, and Genesys Cloud BYOC Cloud. Deployments: on-premises, hybrid, multi-tenant, any, and embedded-library. Sources checked 2026-09-24. On Cisco CUBE, the DTMF relay methods apply to IOS XE 17.6 onwards, and H.323 is not supported from IOS XE Bengaluru 17.6.1a onwards 109.
What remains uncertain
Webex Calling cloud DTMF methods and SIP INFO handling are not covered by the sources below. CUCM 15 SIP trunk DTMF Signaling Method options and MTP rules are not covered by the sources below. Teams Direct Routing DTMF payload type and RFC 2833 expectation are not covered by the sources below. CUCM current support for SIP INFO DTMF is not covered by the sources below. WebRTC DTMF (RFC 7874 and RTCRtpSender insertDTMF) is not covered by the sources below. DTMF with SRTP and media bypass on SBCs is not covered by the sources below.
See also
Related to
- Teams voice routing policies and dial plans — Teams Direct Routing media stack does not support in-band DTMF; SBC must deliver telephone-events toward Teams
- WebRTC in modern contact centers — WebRTC DTMF (RFC 7874 / insertDTMF) not researched in this packet; scored into frontier
Referenced by
- WebRTC in modern contact centers — RFC 7874 makes audio/telephone-event (RFC 4733) mandatory for WebRTC endpoints; the W3C insertDTMF API uses a narrower duration range than RFC 7874.
Sources
- 1In Asterisk, dtmf_mode auto sends RFC 4733 if the other side supports it and in-band otherwise; auto_info sends RFC 4733 if supported and SIP INFO otherwise.res_pjsip - Asterisk Documentation · endpoint > dtmf_mode > auto, auto_info · Checked 2026-09-24
- 2Asterisk res_pjsip endpoint dtmf_mode defaults to rfc4733 and accepts rfc4733, inband, none, info, auto and auto_info.res_pjsip - Asterisk Documentation · endpoint > dtmf_mode · Checked 2026-09-24
- 3Cisco's CUBE technote recommends RFC 2833 toward SIP service providers and both RFC 2833 and KPML toward Unified CM or CME.Configure DTMF Relay on CUBE · Preferred DTMF methods by scenario · Checked 2026-09-24
- 4On CUBE, interworking RFC 2833 with G.711 in-band audio DTMF requires DSP transcoding resources and G.711 end to end; Cisco states the other DTMF interworking cases are handled without a transcoder in flow-through calls.Configure DTMF Relay on CUBE · DTMF interworking section; G711 to RFC2833 requirements · Checked 2026-09-24
- 5Cisco's proprietary cisco-rtp DTMF relay (payload type 121) has been discontinued on CUBE.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, cisco-rtp description · Checked 2026-09-24
- 6Cisco's fix for duplicate digits is to configure dtmf-relay rtp-nte digit-drop on the inbound dial-peer and the out-of-band method on the outbound dial-peer, so CUBE suppresses NTE packets and relays only the out-of-band digits.Configure DTMF Relay on CUBE · Digit-drop section · Checked 2026-09-24
- 7When CUBE interworks rtp-nte to an out-of-band method without digit-drop, the same digit is sent both in RTP and out-of-band and the far end sees duplicate digits.Configure DTMF Relay on CUBE · Digit-drop section · Checked 2026-09-24
- 8CUBE uses payload types 96 and 97 for fax by default, and Cisco advises the asymmetric payload dtmf command when a third-party device's DTMF payload type conflicts.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, payload type note · Checked 2026-09-24
- 9H.323 (and with it the H.245 alphanumeric and signal DTMF methods) is not supported on CUBE from IOS XE Bengaluru 17.6.1a onwards.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, restriction note on H.323 · Checked 2026-09-24
- 10On CUBE (IOS XE 17.6 onwards), SIP DTMF relay methods are rtp-nte (RFC 2833), sip-notify (Cisco proprietary), sip-kpml (RFC 4730) and sip-info, plus G.711 in-band audio, which needs no configuration.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, supported methods list · Checked 2026-09-24
- 11If an endpoint supports none of the DTMF relay methods configured on CUBE, an MTP or transcoder is required.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, fallback/restrictions · Checked 2026-09-24
- 12When more than one out-of-band DTMF method is configured on a CUBE dial-peer, preference runs from highest to lowest in the order the methods were configured.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, dtmf-relay command notes · Checked 2026-09-24
- 13On CUBE, DTMF relay is disabled by default on SIP and H.323 dial-peers except SIP INFO, so a relay method has to be configured.Configure DTMF Relay on CUBE · Section on CUBE supported DTMF methods / configuration · Checked 2026-09-24
- 14CUBE prefers sip-notify or sip-kpml over rtp-nte, and rtp-nte over in-band, when selecting a DTMF method.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, DTMF relay selection/priority description · Checked 2026-09-24
- 15On CUBE, sip-info DTMF handling is always enabled for SIP dial-peers and is invoked when a SIP INFO with DTMF relay content is received.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, sip-info description · Checked 2026-09-24
- 16A Cisco IOS XE gateway can receive SIP INFO with Content-Type application/dtmf-relay carrying Signal and Duration fields and play the DTMF tone on the telephony call leg.CUBE Configuration Guide - Cisco IOS XE 17.6 Onwards - SIP INFO Method for DTMF Tone Generation · SIP INFO Method for DTMF Tone Generation, feature information · Checked 2026-09-24
- 17The CUBE dtmf-relay command accepts up to six methods on a VoIP dial-peer.Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, dtmf-relay method1 [...[method6]] syntax · Checked 2026-09-24
- 18Unified CM inserts an MTP when a SIP endpoint supports only RFC 2833 and the other side needs out-of-band DTMF; the MTP exchanges RFC 2833 packets with the SIP endpoint and out-of-band DTMF with Unified CM.CallManager/ DTMF/ MTP Allocation · DTMF/MTP allocation table and explanation · Checked 2026-09-24
- 19Cisco's CUBE DTMF technote (updated May 2023) states that Unified CM does not support SIP INFO for DTMF.Configure DTMF Relay on CUBE · SIP INFO section · Checked 2026-09-24
- 20The default RTP clock rate for the telephone-event payload is 8000 Hz, which can be redefined when the dynamic payload type is assigned.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.1 · Checked 2026-09-24
- 21RFC 4733 maps DTMF digits 0-9 to event codes 0-9, star to 10, pound to 11, and A-D to 12-15.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 3.2 · Checked 2026-09-24
- 22The telephone-event payload format has no static RTP payload type number; the payload type is assigned dynamically and signalled out-of-band (for example in SDP).RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.1 · Checked 2026-09-24
- 23The final (E-bit) packet for each telephone event and each segment SHOULD be sent a total of three times at the source's update interval.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.5.1.4 Retransmission of Final Packet · Checked 2026-09-24
- 24If no 'events' fmtp parameter is received, an RFC 4733 sender SHOULD assume the peer supports DTMF events 0-15 and no other events.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.5.1.1 Negotiation of Payloads · Checked 2026-09-24
- 25The first RTP packet for a telephone event MUST have the marker (M) bit set.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.5.1.2 Transmission of Event Packets · Checked 2026-09-24
- 26RFC 4733 recommends that a gateway render only the telephone-event payload once it is received, because the concurrent audio may contain spurious tones introduced by compression.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.5.2.2 Playout of Tone Events · Checked 2026-09-24
- 27Genesys lists out-of-band DTMF (RFC 2833) as a required capability for BYOC SIP providers, and says customer endpoints should handle in-band if RFC 2833 is not supported by the PBX or far end.SIP provider configuration information and questionnaire for BYOC Cloud/BYOC Premises trunks · Functionality requirements; RFCs supported (RFC 2833 row) · Checked 2026-09-24
- 28Genesys Cloud external (BYOC) trunks offer three DTMF methods: RTP Events (default, RFC 2833/4733), In-band Audio, and None.External trunk settings · DTMF settings > DTMF Method · Checked 2026-09-24
- 29The Genesys Cloud external trunk DTMF Payload setting accepts 96-127, defaults to 101, and applies only when the DTMF Method is RTP Events.External trunk settings · DTMF settings > DTMF Payload · Checked 2026-09-24
- 30In-band DTMF is recognised reliably only on G.711 (ulaw or alaw); low-bit-rate codecs distort the tones so they may not be recognised.Configure DTMF Relay on CUBE · Common issues: codec distortion / G711 requirements · Checked 2026-09-24
- 31RFC 4730 (November 2006) defines the 'kpml' SIP event package, in which a subscriber sends digit patterns in a SUBSCRIBE and matched key presses are reported in NOTIFY messages.RFC 4730: A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML) · Sections 2 and 4 · Checked 2026-09-24
- 32KPML digit-pattern matching uses longest-match rather than first-match semantics.RFC 4730: A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML) · Section 3.3 · Checked 2026-09-24
- 33RFC 6086 keeps a 'legacy' mode of INFO usage compatible with RFC 2976 for backward compatibility.RFC 6086: Session Initiation Protocol (SIP) INFO Method and Package Framework · Section 3 · Checked 2026-09-24
- 34RFC 6086 attributes industry interoperability problems to legacy INFO usages that require static configuration.RFC 6086: Session Initiation Protocol (SIP) INFO Method and Package Framework · Section 2 · Checked 2026-09-24
- 35If an event lasts beyond the maximum duration field value 0xFFFF, the sender MUST send a packet reporting that maximum without the E bit and continue the event in a new segment.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.5.1.3 Long-Duration Events · Checked 2026-09-24
- 36Cisco documentation classifies RFC 2833 NTE as an in-band DTMF method because the events travel in the RTP stream.disputedCisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, In-Band methods list (rtp-nte) · Checked 2026-09-24
- 37Asterisk documentation describes rfc4733 DTMF as sent out of band of the main audio stream, and Genesys describes RTP Events as out-of-band.disputedres_pjsip - Asterisk Documentation · endpoint > dtmf_mode > rfc4733 · Checked 2026-09-24
- 38PJSIP documentation warns that if the remote side does not support SIP INFO and never replies, the sender treats it as a transaction timeout and disconnects the call.DTMF — PJSIP Project documentation · Sending DTMF · Checked 2026-09-24
- 39PJSIP generates in-band DTMF tones but does not implement in-band DTMF detection.DTMF — PJSIP Project documentation · Overview · Checked 2026-09-24
- 40In PJSIP, sending RFC 4733 DTMF when the negotiated SDP lacks the telephone-event rtpmap returns PJMEDIA_RTP_EREMNORFC2833; any fallback is up to the application.DTMF — PJSIP Project documentation · Overview · Checked 2026-09-24
- 41PJSIP documentation calls RFC 4733 telephone-event the recommended, standards-based DTMF method.DTMF — PJSIP Project documentation · Overview · Checked 2026-09-24
- 42Payload type 101 for telephone-event is a common vendor default rather than a value fixed by the standard, so a far end may legitimately negotiate a different dynamic value.inferredRFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.1 (dynamic payload type), read with Genesys external trunk settings DTMF Payload default 101 · Checked 2026-09-24
- 43Under RFC 6086 a UA supporting Info Packages MUST list the packages it will accept in the Recv-Info header field.RFC 6086: Session Initiation Protocol (SIP) INFO Method and Package Framework · Section 5.2.2 · Checked 2026-09-24
- 44RFC 2833 (May 2000, Proposed Standard) is obsoleted by two documents: RFC 4733 and RFC 4734.Information on RFC 2833: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals · RFC Editor info page, Obsoleted by field · Checked 2026-09-24
- 45RFC 4733 places the duty to avoid double detection mainly on the receiving gateway; the checked sections contain no requirement that a sender mute in-band tones while sending events, so duplicate-digit protection cannot be assumed from the sender.inferredRFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Sections 2.5.1 and 2.5.2.2 (absence of a sender muting requirement in the sections checked) · Checked 2026-09-24
- 46RFC 4733, published December 2006 on the Standards Track, obsoletes RFC 2833.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · RFC 4733 header block (Obsoletes: 2833; Category: Standards Track; December 2006) · Checked 2026-09-24
- 47RFC 6086 (January 2011) defines the SIP INFO method and Info Package framework and obsoletes RFC 2976.RFC 6086: Session Initiation Protocol (SIP) INFO Method and Package Framework · RFC 6086 header block · Checked 2026-09-24
- 48RFC 6086 states that DTMF transport over SIP INFO has been done with proprietary, unstandardised mechanisms.RFC 6086: Session Initiation Protocol (SIP) INFO Method and Package Framework · Section 2 (Motivation) · Checked 2026-09-24
- 49For SIP INFO tone generation the Cisco gateway clamps Duration to 100-5000 ms and uses 250 ms if no duration is given.CUBE Configuration Guide - Cisco IOS XE 17.6 Onwards - SIP INFO Method for DTMF Tone Generation · SIP INFO Method for DTMF Tone Generation, duration parameters · Checked 2026-09-24
- 50For media bypass early media, Microsoft says the SBC should support sending DTMF during the early-media phase to enable IVR and voicemail scenarios.Direct Routing support for media bypass - Microsoft Teams · Scenarios > Inbound call from SBC > Early media · Checked 2026-09-24
- 51The Teams Phone Direct Routing media stack does not support in-band DTMF.Teams Phone Direct Routing: Definitions and RFC standards · Section 'DTMF' · Checked 2026-09-24
- 52The Direct Routing 'RFC standards' list that SBCs must comply with does not name RFC 2833 or RFC 4733, so the page does not itself specify which out-of-band DTMF method or payload type Teams expects.inferredTeams Phone Direct Routing: Definitions and RFC standards · Section 'RFC standards' and subsections (absence) · Checked 2026-09-24
- 53When a PSTN leg delivers DTMF only as in-band audio, the Direct Routing SBC has to convert it to a non-in-band method before it reaches Teams, or Teams IVRs and voicemail will not see the digits.inferredTeams Phone Direct Routing: Definitions and RFC standards · Section 'DTMF' combined with RFC 4733 Section 2.5.2.2 · Checked 2026-09-24
- 54The RFC 4733 payload format for named telephone events is designated 'telephone-event' with media type audio/telephone-event.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.1 · Checked 2026-09-24
- 55RFC 4733 allows a source to choose its own spacing for event update packets and recommends 50 ms.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.5.1.2 Transmission of Event Packets · Checked 2026-09-24
- 56The RFC 4733 volume field expresses power level for tone events from 0 to -63 dBm0.RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.3.4 Volume Field · Checked 2026-09-24
- 57Cisco's Webex Calling Local Gateway configuration uses dtmf-relay rtp-nte on the PSTN-facing dial-peer and on the Unified CM-facing dial-peers.Configure Local Gateway on Cisco IOS XE for Webex Calling · Registration-based LGW: dial-peer 200 (IP PSTN and TDM); Unified CM integration: dial-peers 300 and 400 · Checked 2026-09-24
Documents
Information on RFC 2833: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 4730: A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML)
RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals
RFC 6086: Session Initiation Protocol (SIP) INFO Method and Package Framework
CallManager/ DTMF/ MTP Allocation
Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay
Configure DTMF Relay on CUBE
Configure Local Gateway on Cisco IOS XE for Webex Calling
CUBE Configuration Guide - Cisco IOS XE 17.6 Onwards - SIP INFO Method for DTMF Tone Generation
Direct Routing support for media bypass - Microsoft Teams
External trunk settings
SIP provider configuration information and questionnaire for BYOC Cloud/BYOC Premises trunks
Teams Phone Direct Routing: Definitions and RFC standards
DTMF — PJSIP Project documentation
res_pjsip - Asterisk Documentation
Cite this page
APA
WarmTransfer. (2026, September 24). DTMF methods and interoperability failures. WarmTransfer. https://warmtransfer.net/knowledge/dtmf-interop
BibTeX
@misc{warmtransfer-dtmf-interop,
title = {DTMF methods and interoperability failures},
author = {{WarmTransfer}},
year = {2026},
url = {https://warmtransfer.net/knowledge/dtmf-interop},
note = {Verified 2026-09-24}
}