Re: How to make sip call pass through NAT

[Lai Li]

Hi,

Here are my environment,

one phone with wifi only, and another phone with 4G

Both phone use used test.linphone.org as sip server, and I could make call from wifi or from 4G, the voice is ok during sip call.

But, when I using my flexisip (using the same wifi as the phone, and I also map 5060 to public ip with 10001), I could make call from both side but no voice during sip call.

I wonder the flexisip need to tell the nat open the other voip port sip call?

James

On Wed, 29 Apr 2020 at 18:15, Mikhail Sviridov <address@hidden> wrote:

Can you try to set one phone (which you will call from) as 4G and second one remaining on WiFi?

 

With Best Regards,

Mikhail.

 

Sent from Mail for Windows 10

 

From: Lai Li
Sent: Wednesday, April 29, 2020 1:11 PM
To: Mikhail Sviridov
Cc: address@hidden
Subject: Re: How to make sip call pass through NAT

 

Hi Mikhail,

 

 I used test.linphone.org as my sip server, everything is working fine.

So I think my network provider doesn’t close the voip ports. It might be my flexisip problems.

 

 

 

James

 

On Wed, Apr 29, 2020 at 5:53 PM Mikhail Sviridov <address@hidden> wrote:

Hi James,

 

It could depend on the network you have. It does not work in our network too, but we know from our provider that he closed all VoIP related ports. So we do not have a chance with this network. But… probably we can have some chance with TLS – need to try it out.

 

With Best Regards,

Mikhail.

 

Sent from Mail for Windows 10

 

From: Lai Li
Sent: Wednesday, April 29, 2020 9:22 AM
To: address@hidden
Subject: How to make sip call pass through NAT

 

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?

 

James