Self Hosted SIP Latency Issue

Self Hosting
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1

Hello :waving_hand:

I was working on Livekit telephony directly connected to IMS through my SIP url. I’ve deployed all Livekit services [ Server, SIP, Agent ] on prem. Also created the inbound trunk as well as the dispatch rule. I’ve also a short-code allocated just for this.
I’ve used Google Realtime model and separate STT, LLM, TTS models as well but I’m experiencing a delay (>30s) on the generation of the audio from my Agent. The IMS has a configuration to timeout in 30s of inactivity But my Agent has supposed to be generating the audio before that.
I had 3 successful calls so far, which the agent responded between 7 - 10s after the call get’s picked up. I’ve added many logs to exactly track in which stage it’s failing but even though the call is ended after 30s of inactivity I see the agent logs generating the audio after 10-20 sec after the call has ended.

What could be the issue and how do I resolve this? I’ve asked on ask-ai channel but couldn’t figured it out.

2

Yes, I was experiencing the same issue with the SIP trunk.

3

Bro, let me save you the trouble. In my case, it was all the firewall problem. The RTP ports were not configured as they were expected.

4

I am receiving RTP on both sides and the required ports are open, but I am still experiencing the same issue.

5

That’s Weird. Maybe use the livekit events for more debugging logs and try to see at which stage it’s delaying.