Hello Jaco,
Thanks a lot for adding some context. We do not use asterisk. We wrote our own custom PJSIP based SIP client and talk to Twilio which acts as a SIP proxy. It seems likely a bug/issue on the SIP client/end point side.
Do you know how I could use rport in PJSIP, could not find any documentation. I have seen a setting for disabling STUN however, https://www.pjsip.org/docs/book-latest/html/reference.html?highlight=accountnatconfig#_CPPv4N2pj16AccountNatConfig10sipStunUseE
Could you please provide some details on how you achieved that ? Thanks!
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