VoIP Mailing List Archives

[alan at chandlerfamily...]

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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[demuel at thephinix.org]

What a long detailed list of todos. You certainly can't find that kind of
answers in here.

[jason at jasonjgw.net]

Alan Chandler <alan@chandlerfamily.org.uk> wrote:

[mcampbellsmith at gmai...]

Hi Alan,

I hope you find your answers here as these are the sort of things that
are hard to find on the wiki, which is somewhat outdated in areas. If
you do find your answers, please post them back here for everyone
else.

I am new to FS also, so my comments below may not be 100% correct!

1. Very similar to what I want to have setup as well. Do you have a
static IP address at home. If not, get a dyndns account and setup an
entry there so that your friends/family can register using your dns
name instead of ip address

2. No idea. Maybe try another stun server?

3. Not sure if double-NAT is needed now with the newer builds of
FreeSwitch. Download the latest 1.0.4 to be on the safeside and
compile it again! (I have FS 1.0.4 pre9 and it works I think). As
long as your clients can register remotely you should be okay. I think
FS can work around most home NATs. Make sure you have auto-nat set in
your internal.xml file (I think its this one)

4. SIP is the signaling. RTP is the payload, or voice in your case.
Any transition is done via the SIP signaling. This is how FS can
transfer calls etc or use the media bypass mode by specifying the IP
address where the RTP should be sent, which does not have to be the
same as the signaling. Make sure you enable tracing in the
internal.xml file so you can debug the signaling.

You don't need to take a laptop to your daughters to test this. Use
an internet sip phone like flaphone.com, which works through your web
browser. This will register with an external IP address exactly like
your daughters and save you time traveling. Note that sound isn't so
clear for me using this service, but it helps with debugging.

I also would recommend a sip client on windows like Zoiper, or
CounterPath's X-Lite.. both are free. X-Lite is well known, Zoiper
allows for multiple SIP registrations and comes in a portable version.







On Fri, Aug 7, 2009 at 6:18 PM, Alan Chandler<alan@chandlerfamily.org.uk> wrote:

[alan at chandlerfamily...]

Mark Campbell-Smith wrote: