II.
Channel overview
Reference · livechannel:sip-telephony
SIP / PSTN Telephony overview
Realtime phone transport (SIP/PSTN, inbound + outbound, DTMF, REFER, warm/cold transfer) bridged to WebRTC by the external SIP gateway (livekit/sip). Inbound calls map onto the channels-adapter event->bounded-session-spawn pattern. See docs/research/realtime-voice-agent-stack.md.
Attributes
displayName
SIP / PSTN Telephony
kind
voice
endpoint
sip
persistent
true
description
Realtime phone transport (SIP/PSTN, inbound + outbound, DTMF, REFER, warm/cold
transfer) bridged to WebRTC by the external SIP gateway (livekit/sip). Inbound
calls map onto the channels-adapter event->bounded-session-spawn pattern. See
docs/research/realtime-voice-agent-stack.md.
Outgoing edges
realizes1
- layer:3-transport·LayerTransport
Incoming edges
None.