Telephone Integration issue with firewall

Shakeel Ahmed
Tue, 17 Mar 2026 15:05:15 +0000
Telephony

Hi LiveKit Community,

We are integrating LiveKit SIP with our Asterisk PBX (PBXIP).
According to your documentation, the static IP blocks for SIP are:

143.223.88.0/21

161.115.160.0/19

However, we are receiving SIP traffic from 129.213.97.183 which is outside your published ranges. This causes our firewall/ACL to block the traffic.
SIP message received:

<--- SIP read from UDP:129.213.97.183:20593 --->
BYE sip:3012121211@PBXIP:5060 SIP/2.0
Via: SIP/2.0/UDP 161.115.179.180:5060
From: <sip:3012121255@355ybhu68sw.sip.livekit.cloud:5060>
Call-ID: 366364bb13fa461e0a6acdd404e99a7d@139.64.158.11:5060

Questions:

1. Why is SIP traffic originating from 129.213.97.183 (outside your static range)?

2. Is this a known issue or misconfiguration on your end?

3. Should we whitelist additional IP ranges? If so, what are they?

4. Will static IPs be enforced consistently for all SIP signalling and media?

Please advise so we can whitelist accordingly. (edited)

2 posts - 2 participants