LiveKit SIP + Asterisk: "Media Timeout" Error When Receiving Calls

[haitham-ramadan]

📞 Issue with LiveKit SIP + Asterisk: "Media Timeout" Error When Receiving Calls

Hey everyone,

I’ve configured Asterisk and LiveKit SIP on my server (Docker version), and I'm currently facing a frustrating issue.

When I try to make a call to a LiveKit SIP endpoint:

• The call reaches LiveKit (I see it coming in),
• But the agent does not respond,
• And after a few seconds, I get a Media Timeout error.

Here’s a log snippet from livekit_sip:

sip_1 | 2025-04-21T23:34:19.883Z   WARN    sip     sip/inbound.go:721      Closing inbound call with error {"nodeID": "NE_JKTvPFRB8F4C", "callID": "SCL_AnxXXHcbPbJx", "fromIP": "197.60.191.230", "toIP": "167.71.48.30:5060", "fromHost": "167.71.48.30", "fromUser": "123", "toHost": "167.71.48.30", "toUser": "15105550100", "sipTag": "as0a5b1942", "sipCallID": "[email protected]:5060", "sipRule": "SDR_wchirf4FN4DF", "room": "test_room", "participant": "sip_123", "participantName": "Phone 123", "reason": "media-timeout"}

🔍 I’ve looked through this related issue but haven’t had success:
👉 #231

I’ve double-checked:

• SIP registration is correct
• Asterisk and LiveKit are configured correctly
• Room exists and credentials are right

Still can’t figure out why the media is not being negotiated or received. Any tips?

Thanks in advance 🙏

[dennwc]

The problem is likely that something is blocking RTP traffic between the two services. Are these deployed on the same host?

[KTarun003]

I have the same issue , but I am using livekit cloud. I have verified that RTP packets from asterisk have been sent to the livekit server but i have no way of verifying it if have been recieved there and I am not getting any incoming traffic. I have checked this with Wireshark.

[KTarun003]

My issue has been resolved, my server was in a private network hence I had to add NAT rule and add externip in sip.conf.

Like this:

[general]
externip=<ip>