SIP participant

Mapping a caller to a SIP participant.

note

To create a SIP participant to make outbound calls, see Make outbound calls.

Each user in a LiveKit telephony app is a LiveKit participant. This includes end users who call in using your inbound trunk, the participant you use to make outbound calls, and if you're using an agent, the AI voice agent that interacts with callers.

SIP participants are managed like any other participant using the participant management commands.

SIP participant attributes

SIP participants can be identified using the kind field for participants, which identifies the type of participant in a LiveKit room (i.e. session). For SIP participants, this is Participant.Kind == SIP.

The participant attributes field contains SIP specific attributes that identify the caller and call details. You can use SIP participant attributes to create different workflows based on the caller. For example, look up customer information in a database to identify the caller.

SIP attributes

All SIP participants have the following attributes:

AttributeDescription
sip.callIDLiveKit's SIP call ID. A unique ID used as a SIP call tag to identify a conversation (i.e. match requests and responses).
sip.callStatusCurrent call status for the SIP call associated with this participant. Valid values are:
  • active: Participant is connected and the call is active.
  • automation: For outbound calls using Dual-Tone Multi-Frequency (DTMF), this status indicates the call has successfully connected, but is still dialing DTMF numbers. After all the numbers are dialed, the status changes to active.
  • dialing: Call is dialing and waiting to be picked up.
  • hangup: Call has been ended by a participant.
  • ringing: Inbound call is ringing for the caller. Status changes to active when the SIP participant subscribes to any remote audio tracks.

sip.phoneNumber

User's phone number. For inbound trunks, this is the phone number the call originates from. For outbound SIP, this is the number dialed by the SIP participant.

note

This attribute isn't available if HidePhoneNumber is set in the dispatch rule.

sip.ruleIDSIP DispatchRule ID used for the inbound call. This field is empty for outbound calls.
sip.trunkIDThe inbound or outbound SIP trunk ID used for the call.
sip.trunkPhoneNumberPhone number associated with SIP trunk. For inbound trunks, this is the number dialed in to by an end user. For outbound trunks, this is the number a call originates from.

Twilio attributes

If you're using Twilio SIP trunks, the following additional attributes are included:

AttributeDescription
sip.twilio.accountSidTwilio account SID.
sip.twilio.callSidTwilio call SID.

Custom attributes

You can add custom SIP participant attributes in one of two ways:

  • Adding attributes to the dispatch rule. To learn more, see Setting custom attributes on inbound SIP participants.

  • Using SIP headers: For any X-* SIP headers, you can configure your trunk with headers_to_attributes and a key/value pair mapping.

    For example:

    {
    "trunk": {
    "name": "Demo inbound trunk",
    "numbers": ["+15105550100"],
    "headers_to_attributes": {
    "X-<custom_key_value>": "<custom_attribute_name>",
    }
    }
    }
    caution

    Note the leading + assumes the Destination Number Format is set to +E.164 for your Telnyx number.

Examples

The following examples use SIP participant attributes.

Basic example

This example logs the Twilio call SID if the user is a SIP participant.

if (participant.kind == ParticipantKind.SIP) {
console.log(participant.attributes['sip.twilio.callSid']);
};

Modify AI voice agent based on caller attributes

The following example uses template apps as a starting point. If you completed the Accepting incoming calls quickstart, you can use the agent you created in Step 2: Creating an AI voice agent.

Otherwise, to create a template app, run the following LiveKit CLI command. For example, to create a voice-pipeline-agent-python template app:

lk app create --template voice-pipeline-agent-python

You can find a full list of Public template apps in the LiveKit examples repo.

The following example is based off the voice-pipeline-agent-python example app. The entrypoint function is modified to identify SIP participants.

  1. Add the following line near the top of your agent.py file to import the rtc module:

    from livekit import rtc
  2. Replace the entrypoint function with the following code:

    async def entrypoint(ctx: JobContext):
    initial_ctx = llm.ChatContext().append(
    role="system",
    text=(
    "You are a voice assistant created by LiveKit. Your interface with users will be voice. "
    "You should use short and concise responses, and avoiding usage of unpronouncable punctuation. "
    "You were created as a demo to showcase the capabilities of LiveKit's agents framework."
    ),
    )
    logger.info(f"connecting to room {ctx.room.name}")
    await ctx.connect(auto_subscribe=AutoSubscribe.AUDIO_ONLY)
    # Wait for the first participant to connect
    participant = await ctx.wait_for_participant()
    logger.info(f"starting voice assistant for participant {participant.identity}")
    # Default Deepgram model
    dg_model = "nova-2-general"
    # Check if the participant is a SIP participant
    if participant.kind == rtc.ParticipantKind.PARTICIPANT_KIND_SIP:
    # Use a Deepgram model better suited for phone calls
    db_model = "nova-2-phonecall"
    # Do something here based on SIP participant attributes
    # For example, look up customer information using their phone number
    # If this caller is calling from a specific phone number, do something
    if participant.attributes['sip.phoneNumber'] == '+15105550100':
    logger.info("Caller phone number is +1-510-555-0100")
    # This project is configured to use Deepgram STT, OpenAI LLM and TTS plugins
    # Other great providers exist like Cartesia and ElevenLabs
    # Learn more and pick the best one for your app:
    # https://docs.livekit.io/agents/plugins
    assistant = VoicePipelineAgent(
    vad=ctx.proc.userdata["vad"],
    stt=deepgram.STT(model=dg_model),
    llm=openai.LLM(model="gpt-4o-mini"),
    tts=openai.TTS(),
    chat_ctx=initial_ctx,
    )
    assistant.start(ctx.room, participant)
    # The agent should be polite and greet the user when it joins :)
    await assistant.say("Hey, how can I help you today?", allow_interruptions=True)

Creating a SIP participant to make outbound calls

To make outbound calls, create a SIP participant. To learn more, see Make outbound calls.