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@ariaflowagents/livekit-plugin-transport-sip

v1.0.0

Published

SIP/RTP transport adapter for LiveKit Agents. Run voice agents over traditional telephony with G.711 codec support.

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Maintainers

octalpixeloctalpixel

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Readme

@ariaflow/livekit-plugin-transport-sip

RTP telephony transport for SIP trunk/PBX integrations (UDP signaling + RTP media).

Use this package for:

  • 3CX or PBX SIP trunk integrations
  • SIP over UDP RTP telephony calls
  • G.711 media bridging into AriaFlow voice sessions

Current signaling scope:

  • Incoming INVITE, ACK, BYE, CANCEL, and OPTIONS
  • Final 200 OK only after call bootstrap succeeds
  • Dialog-correct outbound BYE generation with preserved Call-ID and tags
  • G.711 PCMU and PCMA negotiation

Current production boundaries:

  • UDP only
  • No re-INVITE, hold/resume, PRACK, REFER, SRTP, or RTCP yet
  • Best suited today for controlled PBX/SBC deployments rather than arbitrary internet SIP peers

For SIP over WebSocket/WebRTC signaling, use:

  • @ariaflow/livekit-plugin-transport-sip-jssip