asterisk users - Jan 2019 - (NAT) direct media to host on local net when registering from external address

This is going to be a bit of an odd situation, but perhaps might become
more and more common (as mobile phone SIP clients utilize PUSH proxies
instead of the battery draining direct registering with SIP servers).

I have a SIP client which can be on the same RFC-1918 LAN as my
Asterisk server.  Even though it's on the same LAN as the Asterisk
server, it's registration comes from an IP address external to the LAN.
This is because the client is registering to the local Asterisk server
through a SIP proxy server that is external to the LAN.

Is there any way for Asterisk to determine that this is what is
happening and to direct/setup the media session to the client on it's
LAN address?

Put another way, even though the registration comes from an external
(NATted) IP address, I want the media connection to stay within the
LAN.

One solution of course is to add the external IP address of the SIP
proxy -- the address that the client's registration is coming from --
to localnet but that breaks the use-case of the SIP client (which is
mobile) leaving the LAN and having an external IP address.

Ultimately I am hoping there is something in the registration that will
indicate to Asterisk that even though the registration is coming from
an external IP address, the client has an internal IP address that is
on the same network as it is and sets the IP address in the SDP payload
to a local IP address.

I do realize this is fairly non-standard configuration so it might not
be possible.

Any suggestions would be welcome.

Cheers,
b.

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How is your endpoint currently configured in asterisk? Have you tried
rtp_symmetric to see if the endpoint sends audio to asterisk if asterisk
can send audio back to the client?

Alternatively if your SIP Proxy is also a Media proxy you could set the
media_address on the endpoint to be your proxy and let your proxy handle
proxying the media to your endpoint.


On Tue, Jan 15, 2019 at 8:47 AM Brian J. Murrell <brian at
interlinx.bc.ca>
wrote:
>
> Put another way, even though the registration comes from an external
> (NATted) IP address, I want the media connection to stay within the
> LAN.
>
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On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
>  How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer.
> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?
That would require using chan_pjsip wouldn't it?  Not that I am opposed
to trying that.  I need to use chan_pjsip at some point to be able to
authenticate to my SIP provider for SIP SIMPLE anyway.

But will rtp_symmetric really solve the problem?  Isn't the problem the
setting up of the RTP session, so there is no address and port that it
receives from yet?
> Alternatively if your SIP Proxy is also a Media proxy you could set
> the
> media_address on the endpoint to be your proxy and let your proxy
> handle
> proxying the media to your endpoint.
The idea of sending my media out of the LAN (where I have almost zero
latency) and introducing the latency of a round trip to the proxy and
back doesn't seem like a great solution.

Cheers,
b.

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