/live/sessionsCreate session
Create a Live WebRTC session or place an outbound SIP call. Start with the Live prompting guide for session configuration and Telephony and SIP for trunk setup and call monitoring.
Set transport.type to webrtc and supply an SDP offer, or set it to sip
and supply an E.164 destination and trunk credentials. Outbound SIP calling
must be enabled for your organization.
Ringing is limited to 3 minutes and connected calls to 2 hours; these limits
are not configurable in the request.
Returns 201 Created after session initialization. WebRTC responses include an
SDP answer. SIP responses do not wait for the callee to answer. Attach a
sideband connection using session.id to monitor SIP call progress.
Each SIP request creates a new call. If a request times out or the connection fails, retry with caution: the original request may have succeeded, and a retry can place another call.
- IdempotentThe SDK sends
Idempotency-Key, so a retried request is only applied once.
A 400 is returned: Invalid Live session request. A 413 is returned: The Live session request exceeds the supported size limit. A 429 is returned: The session could not start because a rate or quota limit was exceeded. A 500 is returned: An internal error occurred. The response body is plain text. A 502 is returned: The SIP gateway could not initialize the outbound call. A 503 is returned: The session is temporarily unavailable. A 504 is returned: SIP session initialization timed out. The error code is sip_setup_timeout.