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Daily supports connecting standard phone and SIP participants to video calls, enabling telephony integration for contact centers, conferencing, and voice bot applications.

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Prerequisites

  • Paid Daily account — add a credit card in the dashboard.
  • Dial-out approval — fill out the dial-out request form to avoid fraud and misuse. Once approved, set enable_dialout on your rooms.
  • Default limit: max_sip_pstn_sessions_per_room = 5. Contact support to increase this.
We’ve removed the requirement for a paid-support plan for SIP and PSTN dial-in, it is now available for all pay-as-you-go accounts. You just need to have a credit card associated to your domain.

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SIP

Daily supports SIP dial-in and SIP dial-out with the following codecs:
  • Audio: PCMU, PCMA, G722, Opus (default: PCMU/PCMA)
  • Video: H264, VP8 (default: VP8)
If a codec is not supported by the remote party, Daily transcodes the stream (transcoding charges apply — see pricing). Tested with CPaaS vendors (Twilio, Telnyx, SignalWire) and SIP hardware phones (Poly, Cisco). For full SIP setup details, see the SIP guide.

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PSTN phone numbers

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Global dial-in numbers

You can receive PSTN calls without purchasing a phone number. Daily provides global dial-in numbers in 27 countries. Each room gets a unique 11-digit PIN code, generated at room creation and deleted when the room expires. All accounts get US and Canada at no extra charge:
  • US: +1 (209) 503-8039
  • CA: +1 (226) 807-8039
Access to the other 25 countries (including all major European countries, Japan, Singapore, Malaysia, India, China, Hong Kong, Brazil, Mexico) requires a paid support plan. See this page for the full list of countries and numbers.

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Purchasing phone numbers

Daily’s REST API supports purchasing phone numbers in the US and Canada (+1). A single number can handle hundreds of simultaneous calls. A purchased number can can be used for PSTN dialout. However, a number can only be assigned to either PIN dial-in or pinless dial-in, but not both.

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PSTN dial-in methods

Daily supports two PSTN dial-in flows:
  • PIN dial-in — callers dial a phone number and enter a room-specific PIN code.
  • SIP Interconnect and Pinless dial-in — incoming calls trigger a webhook where you can define logic to route the call to a Daily room without requiring a PIN.

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PSTN dial-out

Dial-out checklist (once your account is approved):
  1. Purchase a phone number (skip if dialing a SIP URI)
  2. Enable dial-out on the room
  3. Join the room with an owner token
  4. Dial US/Canada (+1) numbers in E.164 format (e.g., +16505551212). For international dial-out, contact help@daily.co.
Dial out to a mobile or PSTN phone number from within a Daily room:

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DTMF tones

Send DTMF (touch-tone) signals during a call — commonly used for IVR interactions: On a dial-out call, Daily can signal DTMF two ways:
  • RTP telephone events per RFC 2833/RFC 4733. This method works with almost every endpoint.
  • SIP INFO requests that carry an application/dtmf-relay body with Signal and Duration fields.
With the default method: 'auto', Daily uses SIP INFO when the remote SIP peer lists INFO in the Allow header of its answer. If it does not, Daily uses RTP telephone events. Nothing else changes that choice. If you ask for method: 'sip-info' and the far end does not allow INFO, the request returns an error. Receiving DTMF from a PSTN/SIP connection into a Daily room is not yet supported.
Sending more than one toneWhen you send several tones in a single request over SIP INFO, Daily waits a fixed 100ms between tones. You cannot change that gap. digitDurationMs sets the digit duration that Daily signals to the far end. It does not set the gap between tones.There is no queueing or ordering guarantee across separate requests, so two that overlap can interleave their tones.Pass the whole sequence in one request when you can, for example tones: '1234#'. If you have to send digits one at a time, wait for each request to finish before starting the next one and add your own pacing between them.

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Call transfers

Daily provides two transfer methods. Both methods maintain the original SIP/PSTN connection, meaning the initial caller’s connection does not disconnect during the transfer:
  • SIP Call Transfer — keeps the original SIP leg anchored to Daily and redirects the far end to a new SIP or PSTN destination. Both connections remain active and billable.
  • SIP REFER — hands the call off to the originating SIP system. Daily exits the call entirely. A one-time SIP REFER fee may apply.