What a long detailed list of todos. You certainly can't find that kind of answers in here.
I apologize, as my first post to this list, that I ask a detailed set of questions, but I have spend some time looking at all the docs and can't get what I need to do completely sorted in my head. I am definitely one who likes to UNDERSTAND what is happening rather than follow blank recipies, so please bear with me as I try understand all the details. I do understand about networking, NAT etc - but I am new to SIP/RTP and in particular what I think is a double NAT problem
Firstly - what am I trying to achieve:
I am in the UK and have a small home network behind a D-Link DIR-100 Router/NAT/Firewall one of those machines, running Debian Lenny, acts as my main server for everything (and in an earlier incarnation was the firewall/router/nat box too - I only say this is because I had all this working using Asterisk a year or so ago, but with this important difference in configuration). Many of the ports on the firewall are port forwarded to this machine. I have set Freeswitch up on this server to act as a small voip pbx for the home - but MORE IMPORTANTLY - to enable my daughter from her house to talk to us. At my house locally I have a Linksys PAP2T two phone SIP box - and that is working with Freeswitch's default configuration (I set up to be 1000 and 1001 and used all the facilities). I will later add a Linksys SPA 3102 - although I DO NOT intend to use its facility to bridge to the normal phone network.
My daughter, living in another house, also has a Nat box (unknown - its part of her ADSL modem/router/wireless access point) and also has a PAP2T which she will connect to the her network. This will be her phone.
There is a family relation living in Australia who will load up a whatever softphone that we tell him to use. I expect, but don't know, that he will behind a NAT box too.
Later, I have some friends in the USA that I might wish to add it too - especially so that we can hold some teleconferences. They will have a mixture of Windows and MACs, and I will need to recommend softphone clients for them.
I want to set this up as a small private voice network, so anyone can ring anyone else. I will add fancy facilities such as conferencing and voicemail later - I just want to get the basics working first.
Secondly
I installed a stun client on my home machine and ran it against stun.freeswitch.org.
It reported:-
Primary: Independent Mapping, Independent Filter, preserves ports, no hairpin
But I have no idea what this means - I can't find any clear statement via googling for it - how this set of answers maps to the different types of NAT that might be required to get this to all work. CAN SOMEONE ENLIGHTEN me please.
Thirdly
I have set up a sip profile called "double nat" from the recipe in the wiki. This defines the SIP port to be 5090.
However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters house will initiate a connection to my server. Presumably, I have to port forward 5090 from the nat box to my server. IS THAT CORRECT?
I also assume I will have to tell her to use STUN (I believe this is an option on the PAP2T)
Fourthly
If I understand SIP correctly, it just initiates the session and the two end points then communicate directly via RTP. What I don't understand is how does a session transition from SIP to RTP via the connection set up in the the first phase (in terms of passing through the NAT boxes). In particular WHICH OF THE TEST RESULTS from my stun client indicate it will do the right thing. (I am going to take a laptop to my daughters house with a stun client in to test her network this weekend).
Could someone explain please.
Fifthly
Is there a recommended SIP softphone with all the right facilities (STUN support?) that works on MAC and WINDOWS (I only use linux myself).
Apologies for the length of this. I am eager to get the answers so I can use an opportunity this weekend to get it working.
-- Alan Chandler http://www.chandlerfamily.org.uk
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