Outbound SIP INVITE intermittently not reaching Asterisk

Telephony
1

Hello everyone,

We are facing an intermittent outbound SIP delivery issue with LiveKit SIP.

Project:
agentics-voice-system-pbf8ajd4.livekit.cloud

Outbound trunk:
ST_vewBT7KhoXfm
Number: +922138722043
Transport: UDP
Media encryption: disabled

We first used direct PTCL public IP:

182.184.108.36:5072

Then we tested via Playit UDP tunnel:

communication-items.gl.at.ply.gg:45159 -> Asterisk-2:5072

Issue:
LiveKit API/AI Router successfully creates the call and returns success. LiveKit sessions/participants also appear. But in failed attempts, no SIP INVITE reaches Asterisk tcpdump, so Asterisk logs stay blank and the mobile phone does not ring.

In successful attempts, the INVITE reaches Asterisk, routes through from-p2-livekit, dials Telecard, and 2-way audio works.

Recent failed/test call IDs:
PA_eV4jsqLrG3UJ
PA_ATsubkMLGujM
PA_xmS8bQJJZa7z

What we verified:

  • Asterisk listens on UDP 5072.
  • RTP range is 10000-20000.
  • External UDP test to 182.184.108.36:5072 reached Asterisk.
  • Playit tunnel shows verified.
  • Asterisk config works when INVITE reaches it.
  • Issue appears before Asterisk, between LiveKit outbound SIP trunk and destination delivery.

Question:
How can we check whether LiveKit is actually sending the SIP INVITE for a specific outbound call ID/trunk? Is outbound SIP to a UDP hostname/tunnel like Playit supported reliably, or should we only use a direct public IP/static SIP endpoint?

Any guidance on debugging trunk delivery errors would help. Thank you.

2

How can we check whether LiveKit is actually sending the SIP INVITE for a specific outbound call ID/trunk?

Those failed/test call IDs aren’t LiveKit IDs, but the mapping is as follows:

PA_eV4jsqLrG3UJ → https://cloud.livekit.io/projects/p_/telephony/SCL_LwEnnAmQw8sX/outbound
PA_ATsubkMLGujM → https://cloud.livekit.io/projects/p_/telephony/SCL_PLjEKShLZgtA/outbound
PA_xmS8bQJJZa7z -->https://cloud.livekit.io/projects/p_/telephony/SCL_Cfok9Dw3HeHA/outbound

Each of those pages linked above have a button to download the pcap for the failed call, where you can see the SIP INVITE.