asterisk users - Sep 2005 - One-way audio with VPN

On Tue, September 27, 2005 20:22, Alex Lake said:
> I've got a one-way audio problem, but I've looked through a few
> documents on the subject and I'm not sure that it's the same issue.
>
> User A calls a local Asterisk user B via a public SIP gateway
> (voiptalk.org) using (sip:110@siptest.dmclub.net)
>
> B is connected to the Asterisk server via VPN
>
> B is registered (and has successful bi-directional conversations with
> other users on the VPN)
>
> Asterisk correctly forwards the call via SIP and B's phone rings and is
> answered, but B can't hear A
>
> So there appears to be an audio-path blockage from A via Asterisk to B.
>
> Now if A leaves a voicemail message on the asterisk box, that's fine
> (the sound file contains a recording of A's voice!)
>
> Therefore, it looks like the problem is to do with the forwarding of RTP
> packets by Asterisk from A (Internet origin) to B (VPN).
>
> Any ideas?
>
If you're not doing NAT on the SOURCE IP of the A before transferring
across the VPN, it is very likely that B is replying DIRECTLY to A rather
than through the VPN. This will cause B to answer with a different Source
IP than A has initiated the call to, causing the packets to be dropped.

You can easily check this by doing a packet trace on the LAN segment of B...

Good luck!

-- 
Francesco Peeters
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I've got a one-way audio problem, but I've looked through a few 
documents on the subject and I'm not sure that it's the same issue.

User A calls a local Asterisk user B via a public SIP gateway 
(voiptalk.org) using (sip:110@siptest.dmclub.net)

B is connected to the Asterisk server via VPN

B is registered (and has successful bi-directional conversations with 
other users on the VPN)

Asterisk correctly forwards the call via SIP and B's phone rings and is 
answered, but B can't hear A

So there appears to be an audio-path blockage from A via Asterisk to B.

Now if A leaves a voicemail message on the asterisk box, that's fine 
(the sound file contains a recording of A's voice!)

Therefore, it looks like the problem is to do with the forwarding of RTP 
packets by Asterisk from A (Internet origin) to B (VPN).

Any ideas?

Thanks for that. We did track it down to a problem with native bridging. 
In this case, Asterisk assumed that the VPN was publicly accessible - 
but it isn't!

The fix we've found is to setup all VPN-based sip devices with 
canreinvite=no, but I'm not sure if this is the best way to do that.