DTMF Transmission via SIP

This documentation describes how to transmit Dual-Tone Multi-Frequency (DTMF) signaling through the SIP (Session Initiation Protocol) interface using 100ms' REST API. DTMF tones are used for telecommunication signaling over the line in voice-frequency bands to the call control.


The DTMF API provided by 100ms enables the transmission of DTMF tones directly to SIP participants during a call. This capability is essential for interacting with voice response systems, navigating phone trees or for controlling aspects of a session that require numeric input.

Points on Server-Side DTMF Generation

  1. Simplified Client Logic: Offloading DTMF generation to the server reduces complexity on client devices (iOS, Android, Web), which don't need to implement DTMF logic.
  2. Data Integrity: Keeping data in digital form until necessary minimizes the risk of data loss due to packet loss, which can occur when converting digital signals to analog prematurely.
  3. Selective Transmission: Ensures that DTMF tones are only sent to SIP connections, avoiding unnecessary transmission to non-SIP participants.
  4. Vendor Limitations: Not all service providers support server-side DTMF transmission. This could limit flexibility if switching providers or integrating with other systems. However, this is mitigated by 100ms providing a simple client-side implementation for generating DTMF.

REST API Endpoint for DTMF Transmission

Endpoint: https://api.100ms.live/v2/sip/dtmf

Method: POST


  • Authorization: Bearer <AUTH TOKEN> -> Client Auth Tokens are required to access this API; Management Tokens are not compatible with this API. For instructions on generating the auth tokens, please refer to this documentation.
  • Content-Type: application/json

Request Body Example:

{ "digits": ["0", "1"] // Array of DTMF digits }

Request Parameters Description:

digitsarrayAn array of strings, where each string is a character representing a DTMF tone. Valid characters are 0-9, *, #, a, b, c, d.
  • Target Recipients: The DTMF tones will only be sent to all SIP participants present in the call. WebRTC participants will not receive these tones.
  • Transmission Order: Requests are queued and processed sequentially; subsequent requests will not be processed until all previous tones have been sent.
  • Timing: Tones are transmitted at an interval of 100 milliseconds and last for 100 milliseconds. This timing is consistent across all requests.


200 OK - Indicates successful request processing. The body will be empty as the focus is on the action performed rather than data returned.


400 Bad Request - Occurs when there are no SIP participants in the call to receive the DTMF tones.

{ "code": 400, "message": "no sip participants in the call", "details": [""] }

This documentation provides a clear pathway for integrating DTMF transmission capabilities within your applications, ensuring effective interaction with systems requiring numerical input during SIP calls.

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