Telephony

Telnyx is the carrier. Buy a number, connect it to your agent — no third-party SIP trunk setup, no FQDN auth dance. Calls route on-net from Telnyx SIP directly to your agent. For setup steps, see the Quick Start.

Supported

Inbound calls

Inbound calls are routed to your agent via SIP dispatch rules. When someone calls your DID, Telnyx forwards it to the LiveKit SIP service, which dispatches it to your agent based on the rules you configure.

Outbound calls

Use the lk CLI to place an outbound call into a room:

lk sip participant create \
  --room "my-room" \
  --trunk "<SIP_TRUNK_ID>" \
  --call "+15551234567" \
  --identity "outbound-caller"

DTMF

DTMF tones are supported via RFC 2833/4733. Tones are forwarded to your agent as events and can be handled in code.

Call transfers

Cold transfer

Use the stable TransferSIPParticipant API to transfer the caller to another number or SIP endpoint using SIP REFER:

from livekit import api

await ctx.api.sip.transfer_sip_participant(
    api.TransferSIPParticipantRequest(
        room_name=ctx.room.name,
        participant_identity="<CALLER_IDENTITY>",
        transfer_to="tel:+14155550100",
    )
)

See LiveKit’s cold transfer guide for transfer behavior and timeout handling.

Warm transfer

For a stable manual warm transfer, use CreateSIPParticipant to dial the transferee into a consultation room:

from livekit import api

await ctx.api.sip.create_sip_participant(
    api.CreateSIPParticipantRequest(
        sip_trunk_id="<SIP_TRUNK_ID>",
        sip_call_to="+14155550100",
        room_name="<CONSULT_ROOM_NAME>",
        participant_identity="manager",
        wait_until_answered=True,
    )
)

After the consultation, use LiveKit’s stable MoveParticipant API to move the transferee into the caller’s room.

SIP headers

Pass call metadata through to your agent using headers_to_attributes. Header values are mapped to LiveKit participant attributes, available in your agent at runtime.

HD voice

Telnyx supports two HD voice codecs for higher-quality audio on SIP calls:

G.722 — Wideband audio at 16 kHz sample rate. Enabled by default on all Telnyx SIP connections. No additional configuration needed.
Opus — Wideband audio at 48 kHz sample rate. Requires SRTP encryption on both sides — enable OPUS in your Telnyx SIP connection’s inbound codec list and set media_encryption: ALLOW on your LiveKit inbound trunk. Delivers the highest audio quality for voice AI agents.

Both codecs are negotiated automatically during SIP call setup. If the remote side supports Opus, it will be preferred over G.722.

LiveKit on Telnyx (Beta)

Migration

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Plugin Reference

Observability

Reference

For AI Agents

Telephony

Supported

Inbound calls

Outbound calls

DTMF

Call transfers

Cold transfer

Warm transfer

SIP headers

HD voice

​Supported

​Inbound calls

​Outbound calls

​DTMF

​Call transfers

​Cold transfer

​Warm transfer

​SIP headers

​HD voice

Supported

Inbound calls

Outbound calls

DTMF

Call transfers

Cold transfer

Warm transfer

SIP headers

HD voice