Can livekit telephony work without SIP? Using the WSS url? Exotel doesn't support SIP based outbound or SIP REFER based Call forwarding. They support call transfer via WebSocket.

Hey, what is the correct way to manage audio track subscriptions for SIP participants? We need to disable autosubscribe and manually subscribe to desired tracks. Already tried disabling can_subscribe on the SIP participant after creation

await livekit_api.room.update_participant(
            api.UpdateParticipantRequest(
                room=room_name,
                identity=response.participant_identity,
                permission=ParticipantPermission(
                    can_subscribe=False,
                    can_publish=True,
                    can_publish_data=True,
                ),
            )
        )

and manually subscribing via server API

await livekit_api.room.update_subscriptions(
            api.UpdateSubscriptionsRequest(
                    room=room_name,
                    identity=response.participant_identity,
                    track_sids=[target_track_sid],
                    subscribe=True,
                )
            )

But it seems like having can_subscribe False may be disabling any track subscriptions

In Livekit Cloud, we want to listen/export call audio recordings. This is specifically right now for Inbound SIP Trunks. As per documentation, Egress is the way

For LiveKit Cloud customers, Egress is ready to use with your project without additional configuration.

But im not following how to set or use it in the Livekit UI for Cloud. Do we have to set the egress in the code only? like these examples here? I asked in #C088ZNU7QQ5 and this is my present understanding https://livekit-users.slack.com/archives/C088ZNU7QQ5/p1762174619732899

Hi Team, we are experiencing choppy audio on outbound calls. Egress recordings don't have the choppyness, and neither telnyx's recordings. In an earlier message, a similar issue was reported about choppy audio. is it still happening? Thanks!

Hello @refined-appointment-81829 - We closed the agent session and called the delete room API for the below room ID at

Nov 03 22:18:40.450

UTC. But still the room continued to exist and the agent was resumed after 6 or 7 mins. Can you pls check what happened here on LiveKit side logs? Seeing this issue for the first time, hasn't happened before. Room ID -

RM_VZWDb2iHeZMc

Project ID -

p_3tqm7ro6kbs

Hello! I have consistently 15% of my contacts that cannot be heard by a livekit agent. Here are some examples for this project p_46glkdq4l1r

createdAt,room_name,job_id
2025-11-04 10:25:07.475095,0f1471c-1eb2-4660-b7af-ec35cad81053,AJ_sGFW8YRJfcb7
2025-11-04 10:20:17.267128,9258512a-5d03-4cd8-95aa-9c922a50b648,AJ_J5DPwWELqVCG
2025-11-04 09:56:07.332911,4d483127-631a-4bc2-b505-994f5ee00e37,AJ_ZMgYBNXhWmLd
2025-11-04 09:55:15.249522,474ca388-72cb-4699-9b0b-296bb7a4cfc5,AJ_iquvZSWrYYPW
2025-11-04 09:11:32.080618,f8fafdf8-5dfe-401a-be33-93f35e171e5c,AJ_cDCX2Upa5wBb
2025-11-04 09:11:13.223193,574bfa2e-e37c-4d51-8d9a-5cbbc20a2a88,AJ_QckfxRzU5maH
2025-11-04 09:09:20.850155,2bd7d5f1-19b8-45d5-bde3-e27e66bf4ee3,AJ_CcPHbSLzPfyJ
2025-11-04 09:08:46.801538,54193753-d9c5-44b3-8b65-accfec2851ab,AJ_WyTyJdUsKKzX
2025-11-04 09:06:38.920856,50aa0913-0787-4eb0-ab43-aaacee1b0fd8,AJ_u5XbneZcdjRX
2025-11-04 09:01:05.350853,4950ad48-0e56-458e-b567-77f2ddd9242d,AJ_xe2KTvbCVfR

Hello is there any SIP outage that is going on?

I see SIP INVITE timeouts from Twilio -

There was a problem communicating with a specific endpoint of your SIP communications infrastructure. This means there was either a lack of timely response, an error response or an invalid response from your SIP endpoint. This may result in increased call setup times or even failed call depending on the failover configuration for your Elastic SIP Trunk or SIP application. Twilio will make multiple attempts to deliver calls to your endpoint and each failed attempt will have its own notification. The notification will have details about the specific error response and the SIP URI that causes the failure.

cc: @refined-appointment-81829?

👋 Hello, team! I build a voice agent on LK cloud. Works great in playground but I can't seem to figure out how to connect a phone number. I've followed documentation to the letter, tried twilio and telnyx, tried various phone numbers. When I call the number, there is 5-10 silence, then 2-4 rings, then call drops. Any idea what i'm missing?

hey team tryying to figure out why by sip is not working with my sip provider .. the voice of user is getting caputred (confirmed with logs of code ) but voice by bot s not at user end attaching pcap

Anyone any idea why SIP authentication from Twilio works most of the time, but not all the time? PCAP of random failure:

14:40:31.310595 IP ec2-54-172-60-2.compute-1.amazonaws.com.sip > 161.115.178.89.sip: SIP: INVITE <sip:24b873a3-ab48-4f0a-9c8d-7590ca23aacb@4wxwgg2qj6t.sip>.livekit.cloud SIP/2.0
14:40:31.312208 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 100 Processing
14:40:31.351975 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 407 Unauthorized
14:40:31.352024 IP ec2-54-172-60-2.compute-1.amazonaws.com.sip > 161.115.178.89.sip: SIP: ACK <sip:24b873a3-ab48-4f0a-9c8d-7590ca23aacb@4wxwgg2qj6t.sip>.livekit.cloud SIP/2.0
14:40:31.354423 IP ec2-54-172-60-2.compute-1.amazonaws.com.sip > 161.115.178.89.sip: SIP: INVITE <sip:24b873a3-ab48-4f0a-9c8d-7590ca23aacb@4wxwgg2qj6t.sip>.livekit.cloud SIP/2.0
14:40:31.356091 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 100 Processing
14:40:31.358843 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 401 Bad credentials
14:40:31.358943 IP ec2-54-172-60-2.compute-1.amazonaws.com.sip > 161.115.178.89.sip: SIP: ACK <sip:24b873a3-ab48-4f0a-9c8d-7590ca23aacb@4wxwgg2qj6t.sip>.livekit.cloud SIP/2.0

PCAP of it succeeding a few minutes later without any configuration changes:

14:43:24.073062 IP ec2-54-172-60-2.compute-1.amazonaws.com.sip > 161.115.178.89.sip: SIP: INVITE <sip:302365fe-f58f-4b01-8f34-a735f919d486@4wxwgg2qj6t.sip>.livekit.cloud SIP/2.0
14:43:24.074502 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 100 Processing
14:43:24.118803 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 407 Unauthorized
14:43:24.118853 IP ec2-54-172-60-2.compute-1.amazonaws.com.sip > 161.115.178.89.sip: SIP: ACK <sip:302365fe-f58f-4b01-8f34-a735f919d486@4wxwgg2qj6t.sip>.livekit.cloud SIP/2.0
14:43:24.121232 IP ec2-54-172-60-2.compute-1.amazonaws.com.sip > 161.115.178.89.sip: SIP: INVITE <sip:302365fe-f58f-4b01-8f34-a735f919d486@4wxwgg2qj6t.sip>.livekit.cloud SIP/2.0
14:43:24.122531 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 100 Processing
14:43:24.180973 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 180 Ringing
14:43:24.287897 IP 161.115.178.89.sip > ec2-54-172-60-2.compute-1.amazonaws.com.sip: SIP: SIP/2.0 200 OK

Hello we have another one issue with voice of bot we have setup asterisk PJSIp trunk with opus codec. On some calls we have when voice of bot is interrupted. Can you check suggest what can we do for this issue. Link of example call. https://cloud.livekit.io/projects/p_3epz8i5198o/sessions/RM_VsnPnY5xVsx8

Hello Everybody. Have anybody tried bridging call between the Genesys Audio Connector and into a Livekit Agent ? Since Genesys Audio Connector is websocket based I would have thought that this would be possible, but I cant find anybody with experience or discussions about this ? https://help.mypurecloud.com/articles/about-audio-connector/

Hello Everybody, I stugling to connect my livekit ai agent with issbale asterisk based pbx. When user call in my DID/Local IP number the issable accept this then connect with livekit ai agent and user can talk with ai agent. This is my /etc/asterisk/pjsip_custom.conf [transport-udp] type=transport protocol=udp bind=0.0.0.0 [Livekit_AI] type=endpoint context=custom-livekit-dial disallow=all allow=opus aors=Livekit_AI media_encryption=dtls dtls_auto_generate_cert=yes ice_support=yes use_avpf=yes rtcp_mux=yes direct_media=no [Livekit_AI] type=aor max_contacts=1 contact=sip:*************.sip.livekit.cloud:5060 And after the a custon dial plan i make [custom-livekit-dial] exten => s,1,NoOp(--- LiveKit AI Call ---) same => n,Set(ROOMNAME=ai-call) same => n,Verbose(1,Dialing LiveKit room: ${ROOMNAME}) same => n,Dial(PJSIP/Livekit_AI,60) same => n,Hangup() This configaration accept call by astersk and create room in livekit but no audio comes. What is the problem can be and If you know how to achive this please help me. Thank you.

Hello, I have been trying to integrate Livekit sip server with Avaya calls. I am able to successfully make connection to Livekit from Avaya if I directly confiure the phonenumber to establish SIP TRUNK at the start of the call. However, when i try to transfer call to livekit sip in the middle of our IVR call using a vdn which is meant to establish sip trunk to livekit server, the request go to sip server, but livekit sip server rejects it with “no audio in sdp”. The Sip inbound trunk and dispatch rule seem to be working fine. If I want to be able to establish the connection in middle of the call, how can I achieve this?

sip:******.sip.livekit.cloud => instead can we do via IP address ?

Hello, Our team is working on setting up LiveKit on EKS, and we're specifically looking into deploying the LiveKit SIP server. Has anyone had experience with deploying the LiveKit SIP server on EKS? I have a few questions related to the ports that need to be exposed, as our DevOps team has raised some security concerns regarding this.

Hello there. My SIP to livekit was working fine up but since yesterday livekit no longer answers. What could have changed?

In the create sip participant api call, we have set ringing_timeout to 45 seconds. This works, but we have just one problem. It seems that on the SIP level, sometimes the 487 is acknowledged by client (livekit), but a lot of times it is not. Two screenshots are attached to show each situation. So asterisk ends up having to resend the 487 message. What could be the possible reasons/fixes for this? Thank you!

Hello, I got a number from twilio, setup the sip trunk, my freeswitch server receives the call and tries to bridge to livekit. but livekit returns 503. Here is the summary of the problem Phone number +19793169351 configured in LiveKit SIP trunk returns "503 Service Unavailable" when called. Other phone numbers on the same FreeSWITCH server, using the same LiveKit trunk and dispatch rule, work correctly. ## Observable Facts ### The Failing Number - Number: +19793169351 - Provider: Twilio Elastic SIP Trunk - Caller origin: +15106049930 - Response: 503 Service Unavailable from LiveKit (within 1ms) ### Working Numbers (Examples) - Number: +902162351219 (Provider: Verimor) - Number: +902162125830 (Provider: Wespoke) - Same response flow: 100 → 180 → 183 → 200 OK → Call connects ### Infrastructure (Same for All) - FreeSWITCH server: 91.98.71.57 - LiveKit trunk: ST_CxLmHYdYMLRy - LiveKit dispatch rule: SDR_xZ8LxKK2tDtd - LiveKit endpoint: 567c09dlu6y.eu.sip.livekit.cloud:5060 (UDP) - Agent infrastructure: Running and working ## SIP Message Details ### INVITE FreeSWITCH Sends to LiveKit (Failing Number) INVITE sip:19793169351@567c09dlu6y.eu.sip.livekit.cloud:5060 SIP/2.0 From: "UJDDGMFMON" sip:freeswitch_uz@567c09dlu6y.eu.sip.livekit.cloud To: sip:19793169351@567c09dlu6y.eu.sip.livekit.cloud:5060 X-DID: 19793169351 X-From: +15106049930 X-Called-Number: 19793169351 X-Caller-Number: +15106049930 X-Call-ID: 37e84faa-ad2c-4786-aabc-5e395d12f5e2 X-FreeSWITCH-Server: http://91.98.71.57:8888 X-FreeSWITCH-UUID: 37e84faa-ad2c-4786-aabc-5e395d12f5e2 ### LiveKit Response (Failing Number) SIP/2.0 503 Service Unavailable Reason: Q.850;cause=16;text="NORMAL_CLEARING" LiveKit Configuration SIP Trunk Numbers Array "numbers": [ "+19793169351", "19793169351", "9793169351", "+902162351219", "902162351219", "2162351219", "+902162125830", "902162125830", "2162125830" ] SIP Trunk Settings { "allowedAddresses": ["93.174.164.26", "91.98.71.57", "10.1.99.0"], "authUsername": "freeswitch_uz", "authPassword": "LiveKit2024SecurePass!", "headersToAttributes": { "X-DID": "did", "X-From": "from_number", "X-Called-Number": "did", "X-Caller-Number": "from_number" } } Dispatch Rule { "rule": { "dispatchRuleIndividual": { "roomPrefix": "call" } }, "trunkIds": ["ST_CxLmHYdYMLRy"], "metadata": "{\"dispatch\":\"agent\",\"trunk\": \"freeswitch-93.174.164.26\", \"assistant_id\": \"${assistant_id}\", \"phone_number\": \"${from_number}\", \"call_id\": \"${call_id}\", \"did\": \"${did}\"}", "roomConfig": { "agents": [ { "metadata": "{\"assistant_id\": \"${assistant_id}\", \"phone_number\": \"${from_number}\", \"call_id\": \"${call_id}\", \"did\": \"${did}\"}" } ] } } FreeSWITCH Configuration Failing Number Dialplan Wespok e1|\+19793169351|19793169351|9793169351)$" action application="set" data="sip_h_X-DID=19793169351"/ action application="set" data="full_did=+19793169351"/ <action application="system" data="/usr/local/freeswitch/scripts/call-workaround.sh '${full_did}' '${uuid}' '${caller_id_number}' '$${freeswitch_server_url}' '$${sip_media_shared_token}'"/> <action application="bridge" data="{sip_h_X-DID=19793169351,...}sofia/g ateway/livekit_udp/19793169351"/> /condition Working Number Dialplan (Example) <condition field="destination_number" expression="^(gw\+VERIMOR|gw\+902162351219|902162351219|2162351219)$"> action application="set" data="sip_h_X-DID=902162351219"/ action application="set" data="full_did=+902162351219"/ <action application="system" data="/usr/local/freeswitch/scripts/call-workaround.sh '${full_did}' '${uuid}' '${caller_id_number}' '$${freeswitch_server_url}' '$${sip_media_shared_token}'"/> <action application="bridge" data="{sip_h_X-DID=902162351219,...}sofia/ gateway/livekit_udp/2162351219"/> /condition What We've Verified - ✅ Number is in trunk's numbers array (3 formats) - ✅ Source IP (91.98.71.57) is in allowedAddresses - ✅ Authentication succeeds (no 401/407) - ✅ Trunk ID matches dispatch rule trunkIds - ✅ Backend assigns assistant successfully - ✅ FreeSWITCH sends INVITE to LiveKit - ❌ LiveKit immediately returns 503 Question What causes LiveKit to immediately return 503 Service Unavailable for this specific phone number configuration while accepting other numbers on the same trunk with the same dispatch rule?

We have some users who we want to transfer to a human agent immediately, instead of sending them to our LiveKit agent. I tried setting that up by creating a

TransferSIPParticipantRequest

after

ctx.connect

and

ctx.wait_for_participant

, but getting: `livekit.api.twirp_client.TwirpError: TwirpError(code=failed_precondition, message=twirp error unknown: can't transfer non established call, status=412)`Any ideas how to fix this? Not sure how to wait for the call to be "established". Code:

async def entrypoint(ctx: JobContext):
    logger.info(f"connecting to room {ctx.room.name}")

    await ctx.connect()
    user_participant = await ctx.wait_for_participant()
    logger.info(
        f"participant joined, participant={user_participant}"
    )


    session_data, init_job_metadata = await get_init_userdata(
        ctx, user_participant
    )

    # Cold transfer if needed
    if session_data.transfer_on_callback:
        logger.info("transfer_on_callback is true, initiating cold transfer")
        async with api.LiveKitAPI() as livekit_api:
            transfer_to = "tel:" + env("TRANSFER_PHONE_NUMBER")

            transfer_request = TransferSIPParticipantRequest(
                participant_identity=user_participant.identity,
                room_name=ctx.room.name,
                transfer_to=transfer_to,
                play_dialtone=False,
            )
            logger.debug(f"Transfer request: {transfer_request}")

            await livekit_api.sip.transfer_sip_participant(transfer_request)
            logger.info(
                f"Successfully transferred participant {user_participant.identity} to {transfer_to}"
            )
        return

Hi team, We’ve been running SIP ingress for several months without issues, but suddenly all inbound SIP INVITEs we send stopped getting any response. (No configuration change from our side, in btw) • Our proxy (Kamailio at 34.64.81.145, private 10.178.0.22) sends INVITEs to 150.230.105.170:5060 (sip:07080951261@4tk4mu08t2g.sip.livekit.cloud). • We retransmit 10 times (500 ms → 4 s intervals) but receive no SIP responses (no 100/180/200/error). After ~15 s our carrier CANCELs, so the call ends with a 487 on our side. • Checks on our end: ◦ Ping to 150.230.105.170 succeeds; UDP 5060 is reachable (nc -u). ◦ GCP VPC firewall not the issue. ◦ Incoming OPTIONS from our carrier reach us normally, so inbound handling on our side seems fine. Happy to share PCAPs or Kamailio logs if needed.

Hello everyone, @refined-appointment-81829, I’m facing some issues with the inbound SIP integration between Asterisk and LiveKit SIP. I’ve created the

inbound_trunk.json

file with the correct IP addresses and numbers, along with the dispatch rule and agent configuration. On the other side, my Asterisk setup is properly configured. When I originate a call from Asterisk, it keeps ringing continuously. However, on the LiveKit side, my agent gets triggered and starts generating streams after the call starts. In the PCAP and SIP data, there is no 200 OK / ACK message, and from the Asterisk side, the call keeps ringing. I’m not able to pinpoint the exact issue — I’m confused because if the SIP session isn’t accepted, how is the agent getting triggered?

Hi everyone, We self-host our LiveKit service and are currently encountering the

opus: invalid argument

bug, which was fixed in this pull request. As far as I understand, the fix has already been tested and deployed to LiveKit Cloud. However, it doesn’t seem to be available in Docker hub, the latest image there was released about 5 months ago. Could someone please advise when a new Docker image release is planned?

Been trying for the last 4 hours, still getting 486 using the lastest sip 1.2.0 , tried yml etc

Hello everyone, I'm using self-hosted livekit for doing outbound calls but currently even if the call goes to voice mall the agent will be treating that as the customer is speaking, i did search livekit documentation seems that this issue have not be solved. Anyone faced this issue before?

Hi everyone I’m new to SIP integration. My code used to work fine with Twilio, but it’s not working with the Airtel SIP details. Could you please help me understand how to connect the Airtel SIP trunk with LiveKit?

Hello everyone, I’m making an outbound call with twilio and when the client respond I bridge to livekit and send Invitation to it and according to pcap it responds with ACK .. but the twillio call ending without hearing the agent respond .. and when Iook in twilio, I saw the below image I also make sure that the room that I addressed was open and the dispatch rule is defined as following: { “sipDispatchRuleId”: “SDR_3xVKVYuEby5u”, “rule”: { “dispatchRuleDirect”: { “roomName”: “${ROOM_NAME}” } }, “name”: “auto-spawn-ai-agent-on-sip-join”, “metadata”: “{\“version\“: \“1.0.0\“, \“description\“: \“Automatically spawns AI voice agent when SIP participant joins room\“}”, “roomConfig”: { “agents”: [ { “agentName”: “nowlun-voice-assistant” } ] } } the sip inbound trunk defined as following: { “sipTrunkId”: “ST_StxfqYvYn2ko”, “name”: “Twilio Inbound Trunk”, “numbers”: [ “+16362461815” ], “allowedAddresses”: [ “0.0.0.0/0” ] }

I successfully integrate the sip with freepbx.But the issue is that the livekit trunk is reachable and sudennly unrechable again and again and aftre 1 or 2minutes it reachable again. This is working like a loop. Not constently reeachable like other trunk. How can I solve it? On the log i see freepbx sent option method request again and again the livekit do not sent 200 ok. This is the probeble resone. Aftre 1 or 2 minutes it sent 200 and the registration is okay. Please help me fix it. And connect the trunk constently.

[SOLVED] Hey guys, I'm trying to set up an outbound trunk with LiveKit Cloud and Twilio Elastic Trunk. I spent hours of debugging various issues and unclear errors caused by LiveKit documentation. Currently, I'm facing an issue with this error:

lk sip participant create participant.json

Using default project [test]
SIPStatusCode: 403
SIPStatus: Forbidden
rpc error: code = PermissionDenied desc = unexpected status from INVITE response: sip status: 403: Forbidden

I just go through the documentation step by step but alway run into some random issue caused by configuration that is not covered in the LiveKit docs. I think there is an issue with the ACL of IPs, but LiveKit Cloud doesn't have static outbound IP addresses and Twilio doesn't allow to create

0.0.0.0/0

Have you solved this issue somehow? Thanks for any help in advance!