We are testing the Remote Phone as supported in SP2.
The test platform is based on a Syspine base unit and Aastra phones.
The hardware VPN is a pair of Linksys VPN routers (RV082 Central site, RV042 remote)
with both at latest FW releases.
VPN tunnel has been confirmed using basic IP routing techniques (ping,tracert)
and so connectivity looks correct.
Everyting seems to work perfectly except that established calls only are doing one way audio.
That is voice does not come back from local user - that is the remote user phone can not hear the user talking at the central site.
Also if we initial a call to an outside line from the remote phone the same issue occurrs.
Strangley enough an Intercom call from remote phone to other internal phone user works perfectly.
The VPN is setup is very basic as both WAN interfaces have real IP address.
Both Central site and remote of course are private networks.
I have tried combinations of NAT Traversal with no improvement.
CLearly this seems to be an issue with the RTP protocol not flowing - at least one way.
Have been unable to find a hit directly on this issue now that it is supported in SP2.
I of course have to belive it is an issue with Linksys VPN tunnel.
OK I did something stupid - in that we where testing to a non-default gateway device.
Had that covered using overrides to this device - except I changed test phone at central site and forgot to change that one. So - the RTP traffic could not respond back to the remote site.
That makes perfect sense - and once I can ping the phone I was calling from the remote site - then of course it worked.
The problem still remains that I can not get two way audio when calling out to a POTS line.
THe base unit (a Syspine with internal ATA's) makes the call and connects but only one way audio is happening. So I suspected that the internal ATA's are of course also IP devices.
So - they are not getting the right DFGW - and so the RTP traffic doesn't get back to the remote. I have to reboot my unit to get those to take over - so will do that and test later.
It is obvious when the ATA is external - but not so obvious on thebuilt in units.
So the moral of the story is to be careful with routing when using more than one router on a network. I knew that of course - but figure this might be helpful to others.
In many cases with VOIP systems one might have multiple INternet connections.
How many phones do you have at the remote site? If you have multiple phones, is it happening with all of them or just one of them? Have you tried resetting the phones to factory default? And do they have the latest firmware?
I don't know what kind of functionality is built into the Linksys VPN routers, but I'd imagine that it's not particularly function-laden in terms of diagnostics. What you can try to do to see whether the traffic is making it back across the VPN tunnel is to set up a hub that connects to a PC running Wireshark, a remote site phone, and a connection to the Linksys. Try making and receiving calls and seeing if you can see SIP and RTP traffic flowing back from the main site. If you've got a way to watch the traffic from the router, that would make life easier.
Also, do you have a different brand of VPN routers that you could try? I've seen it happen where sometimes between all of the encapsulation/decapsulation, firewalls, and NAT translation that sometimes the SIP and RTP traffic can get mucked up traveling through the internet. If that's the case, and it'd take a good deal of in-depth study to find out, you'd have to contact Linksys/Cisco and let them know. Not saying that that's what is happening, but it's a possibility.