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[kevinlong206 at gmail.com]

Hello,

I have a somewhat confusing use case. We use a mobile voip app and our users connect to our PBX via a public IP of our firewall which port forwards to asterisk (TLS and SRTP ports). Works fine.

Sometimes however, our users are also connected to our VPN (LT2P/Ipsec) which is served by the same firewall that our PBX sits behind at the datacenter.

In this case, most often the calls go through but there is no audio.

I believe that asterisk “thinks” in this case that the IP of the clients, to send RTP traffic to ,t is the firewall’s IP, rather than the IP that the VPN server assigned the client device.

Does asterisk send RTP traffic to the IP which is in the IP headers of the SIP REGISTER , or can a client “specify” it’s truly reachable IP ?

I hope this makes sense.

Regards,

Kevin Long




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[duncan at e-simple.co.nz]

HI Kevin

Is your VPN set as a localnet? The externip only tends to cope with the
firewall address. If you put the VPNs in the localnet lists then it
won't use NAT to find them.

In answer to your question, the SIP session description in the call
setup has the IP for media for both parties, which is where Asterisk /
client will send RTP to respectively . You can look at this using
tcpdump. c= is what you are looking for.

Some formal examples

https://tools.ietf.org/html/rfc4317

Cheers Duncan

On 24 Nov 2015, at 10:01, Kevin Long wrote: