PJSIP Occasionally Stops Processing Calls to Completion

Isn’t that the case here? To me Asterisk defines an endpoint that communicates with the ITSP’s server and Asterisk is behind NAT in this case, or I have misunderstood this part of the configuration.

Think of it this way:

Force rport means you don’t believe the address in the via header. Any ITSP directly on the real internet, will generate a Via header with its true public address, and one would hope any competent ITSP would do the equivalent of Asterisk externhost and transmit the predicted public address (or it should be setting rport itself). One wouldn’t expect the customer’s NAT to change the IP in the outbound direction.

Similarly for comedia. You would expect an ITSP to send SDP with the actual address needed to reach it.

It controls functionality when talking to a specific endpoint, but the options generally reflect interaction with the endpoint. In the case of NAT options, those would be in relation to the remote endpoint itself. “When talking to endpoint A I want to ignore their RTP address, and use where media actually comes from” for example.

I think I got it. The ITSP’s endpoint won’t usually be behind NAT and therefore these options do not matter in these cases.

If I had a second private PBX or a remote phone and theses boxes would be behind NAT, then it would matter.

Indeed, that is correct.

I guess you are saying, that if the correct signalling and media addresses are given, then there’s no need for further settings and connected media stuff is also not needed. Depending on the NAT table timings of the router some special outbound NAT and forwarding rules may be necessary to keep the ports open for a sufficient amount of time.

Time to conjugate that through all possible variants for the accounts I have and write that up…

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