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How to make sip call pass through NAT

From: Lai Li
Subject: How to make sip call pass through NAT
Date: Wed, 29 Apr 2020 14:17:48 +0800

Hi All,

I am using flexisip as my sip server and I have two mobile phone, I am trying to make sip call from one phone(with linphone app) to another phone(with linphone app).
It works fine when both phones are under the NAT, but when a phone connected to internet(without nat) by 4G, I can't here any voice/video after sip call has been created.
But it works fine when I use test.linphone.org as my sip proxy. I think there must be something that I don't know need to be added in flexisip.conf.
How do I fix my configuration? or is there anything I need to do?


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