SIP dial-in (audio only)
Single SIP endpoint
Set thesip room property to enable SIP dial-in:
sip_uri.endpoint — the SIP address your client dials:
sip to null to disable SIP on a room.
Multiple SIP endpoints
Usenum_endpoints to provision multiple SIP dial-in addresses. Each SIP and PSTN connection counts toward the max_sip_pstn_sessions_per_room limit (default: 5).
sip_uri.extra_endpoints with additional SIP addresses:
SIP dial-in (audio and video)
Setvideo: true to enable video SIP dial-in. You can also specify codecs explicitly:
Dynamic display names
display_name is set at room creation time. To pass a name at dial time instead, append it as a URL-encoded query parameter on the SIP URI:
encodeURIComponent to encode the name. If x-daily_display_name is not passed, the room-level display_name is used.
Testing SIP dial-in
Below describes the SIP dial-in flow for a room with one SIP endpoint. For multiple endpoints, the flow is the same but with additionaldialin-ready events and SIP URIs.
- A session starts and the SIP worker registers with the SIP network.
- The
dialin-readyevent fires for each registered SIP URI. - After
dialin-ready, your SIP client dials thesip_uri.endpoint.
- Each
sip_uriendpoint can only be used by one SIP client at a time. - After a disconnect, the same
sip_urican be redialed. - When the Daily session ends, all SIP connections are terminated.
SIP dial-out
A Daily room can dialout to a one or more"sipUri". You can initiate dial-out from the client SDK or REST API. See the overview for details on dial-out prerequisites and supported codecs.
Dial out audio-only:
sip:. You can also use startDialOut() from the client SDK.