One audio on dual stack 4G networks

[dbbrito]

One audio on dual stack 4G networks

I have a problem with FS only when my customers use 4G on a particular operator that uses ipv4 and ipv6, always calls are only with 1 audio, the destination can never hear the source! And in FS it always shows the two registration IPS, at first I thought it was because my FS is dual stack, I disabled the ipv6 profile, removed ipv6 from the server and even registering now only with ipv4 continued with the error! I returned the ipv6 profile to the server and started the tests again. The solution is as follows, I entered the cell phone and went to the APN of the mobile network and it was marked IPV4 and IPV6, I selected only IPV4 and that was it, the problem was solved and all the calls are ok, the problem is that there is no way to pass this solution for all customers, as there are customers who do not have any experience to make this change, so I ask, is there any way to configure FS to resolve this issue without having the customer make changes to their device?

Thank you all!

 

[gflow]

Have you tried SIP TCP?

 

[dbbrito]

Yes, I'm using transport=tcp

 

[Bifur]

When you removed ipv6, did you restart free switch? I enabled tls 5061 the other day and it wouldn't actually connect properly until I restarted free switch. Could be similar issue. If you are only allowing ipv4 on the server then the client should only be able to connect via ipv4.

 

[dbbrito]

When you removed ipv6, did you restart free switch? I enabled tls 5061 the other day and it wouldn't actually connect properly until I restarted free switch. Could be similar issue. If you are only allowing ipv4 on the server then the client should only be able to connect via ipv4.

yes, the switch has been restarted!

 

[dbbrito]

I just discovered a detail, if I have a softphone registered to the wifi, all calls are ok, if I just deactivate the wifi without closing the zoiper it registers normally on the 4G and we only have 1 audio, if I close the zoiper and open it again calls are ok with both audios.
And this problem is not in Zoiper, I just tested it on linphone, and gswave too, that is, when switching from wifi to 4G I have to close the softphone and open it again to work both audios! The problem is to teach all this to the client, would there be any configuration in FS that would solve this issue?

 

[hfoster]

You can make it better by tweaking register timers, so that it registers more often. You'll also fine they go to sleep and ignore invites without implementing something to trigger Google Firebase or Apple Push Notifications on SIP messages too.

It's incredibly painful using SIP on a 4/5G network without some sort of tunnel to bypass NAT though, which I think is what all the premium clients do.

 

[dbbrito]

You can make it better by tweaking register timers, so that it registers more often. You'll also fine they go to sleep and ignore invites without implementing something to trigger Google Firebase or Apple Push Notifications on SIP messages too.

It's incredibly painful using SIP on a 4/5G network without some sort of tunnel to bypass NAT though, which I think is what all the premium clients do.

The biggest problem is when the client switches from the wifi network to 4G without leaving the softphone, this process is what I still don't understand how to get around. Because if he goes out and back in the zoiper again, both audio channels work. Are you talking about enable-timer=false?

 

[hfoster]

The client is 'supposed to' re-register/reinvite if a network change happens. You can always experiment with a SIP keep-alive option, I believe there's one in Zoiper, not sure about this 'enable-timer'. You might be best contacting them for advice on it.

Don't expect anything seamless on SIP though whilst switching networks, it's always going to be a bit of a mess unless you tunnel everything.

 

[dbbrito]

The biggest problem is that I've already tested Zoiper, Linphone and GSWave, and they all happen the same thing, so I think it might be some FS variable.

 

[hfoster]

They're all free apps too, so there's no tunnelling proxy in the middle.

Linphone does have a FOSS solution to this called FlexiSIP that you can deploy, though I never have done myself. It acts as a man in the middle to sort out NAT traversal, media relay, and the mandatory push notifications that modern mobile phones need to wake up the app to process SIP invites.