[asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Victor Villarreal]

Hi Ernie,

When one-way audio appear (no matters if there is a VPN or NAT server on
the diagram) I simply :

* Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x'
on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want
to debug.

* Make a test call and replicate the issue.

* Stop debug with 'sip set debug off'.

* Follow the SIP conversation. Verify that the INVITE message has the
correct IP on the contact field and any other related fields.

* On SDP handshake, verify that the ports where the sound is send, is
correct.

Normally, one-way audio is faced when one audio stream (example the called
audio) is send to the correct IP and Port destination, on the other audio
stream (example the caller audio) don't.

Last, if Asterisk is 'behind' another server, you need tell Asterisk what
is the external IP so it can inform this IP to your clients.

If you dont want to follow the SIP conversation on plain text, you can make
a packet capture on the Asterisk server, instead of SIP debug.

El 19 abr. 2017 16:38, "Mark Wiater" <mark.wiater at greybeam.com> escribió:

> On 4/18/2017 7:40 PM, Ernie Dunbar wrote:
>
>> Server network: 192.168.0.0/24
>> OpenVPN network: 10.8.0.0/24
>> Asus network: 192.168.1.0/24
>>
>> The Asterisk SIP registration appears to be responding properly to this -
>> this is what I see when I do a 'sip show peer' for an Aastra phone that's
>> connecting through the VPN (Asterisk output is truncated):
>>
>>   ToHost       :
>>   Addr->IP     : 10.8.0.6:5060
>>
>
> If the Asus network is 192.168.1.0/24, and the phone is registering as
> 10.0.8.6, it looks like NAT is taking place. Would your asterisk server
> know how to route traffic to 192.168.1.0/24?
>
> I've always used site-to-site OpenVPN tunnels where the vpn's terminate on
> the gateway for both the phones and the asterisk server. I've always had
> rock solid connections between phones and Asterisk.
>
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