Problem when implimenting TLS - Asterix & VoIP.ms

[FreddyK]

Hey there,
I’m currently setting up an Asterisk 20.2.0 on a Ubuntu server 20.04.5 on a Virtual Machine (vmware) .
I’ve setup the server and then registered it with VoIP.ms.

I use Microsip to make the calls on my PC.
(second extension is on a different VM)

When I use the Transport-UDP-NAT lines configuration in my pjsip.conf, everything works fine.
I can call between differents extensions and I can call to a cell phone no problem…

When I try to implement TLS, i’m getting some major problems.

After changing the the configuration in my pjsip.conf to Tranport TLS and setting up the Encryption in VoIP.ms and Microsip, I can call to a cell phone, but there is some heavy distortion on the line for the cell phone user , but eveything is ok on my end. I can hear the cell phone user loud and clear…But not the other way around.

Also, I can call between my extensions, there is connection…but no sound what so ever…
I get no Error Logs in my CLI, eveything seems to be operating smoothly at first glance.

The connection on Microsip is orange instead of green on all my calls after the TLS implementation:

[IMG]https://imagizer.imageshack.com/img923/1572/FqYECO.jpg

When I check the status on VoIP.ms , encryption seems to be implemented and registered fine.

I’m setting up this server as part of a school project, and I use the same type of configurations files my peers are using, but it seems that I’m the one having these kind of troubles…
I’m new to this, I tried reading past threads on the forum to help me troubleshoot, I also tried contacting VoIP.ms…
Can you guys help me out ?

Thank you for your time.
Much appreciated.

[david551]

@FreddyK

I can call to a cell phone, but there is some heavy distortion on the line for the cell phone user

The TLS part has definitely worked, and if this was an encryption related problem with SRTP, it would result in meaningless noise not distorted audio. This sounds like network quality problems.

@FreddyK

The connection on Microsip is orange instead of green

I have no knowledge of MicroSip, so this is not useful to me.

@FreddyK

Also, I can call between my extensions, there is connection…but no sound what so ever…

This could happen on TLS if you were relying on the router to compensate for NAT, as the router would no longer be able to see inside your signalling. Turn off SIP ALG and configure Asterisk to provide the correct public addresses.

[FreddyK]

Hey David, thank you for your answers. Much appreciated.

The TLS part has definitely worked, and if this was an encryption related problem with SRTP, it would result in meaningless noise not distorted audio. This sounds like network quality problems.

On the end of the cell phone user, it’s like “popcorn” on the line, like some wind noise when you ride in your car and the windows are down and it make the crazy wind sound…
On the endpoint end, I hear the cell phone user perfectly.

This could happen on TLS if you were relying on the router to compensate for NAT, as the router would no longer be able to see inside your signalling. Turn off SIP ALG and configure Asterisk to provide the correct public addresses.

I just double checked, SIP ALG is turned off on my router. The right public address is also correct in my pjsip.conf .

I’m starting to go crazy on this one. Can’t figure it out.