Robert Brockway wrote:
> On Tue, 6 Apr 2010, Edward Ned Harvey wrote:
> 
>> If we only cared about the client/server model, IPv4 and NAT would be just
>> fine, now and for decades to come.  The real value-add of IPv6 is peer to
>> peer.  But in order for that to be logistically possible, a client needs to
>> know the IP address of another client, and one thing I know for *dang* sure
>> is that you don't want to read a 128-bit IPv6 address over the phone to
>> somebody typing it into their video conference TV at the other end using
>> their remote control.
> 
> Hi Edward.  Isn't this precisely what SIP is for?  SIP initiates the 
> connection (hence the name) but the connection itself is carried on by 
> other protocols (like RTP).   SIP isn't peer to peer but the connection 
> subsequently negotiated can be.

You need to tell SIP where to connect to.  Some sort of SIP service.
Then you can "dial" anyone else on the same service.  SIP will negotiate
the media (RTP) and any other features.

If you run a SIP client (xlite), you can type in the IP address of the
end peer (p2p), or the SIP directory of your choice (Client/Server like
"Free World Dialup")

SIP doesn't really work over NAT.  The recommendation in the Asterisk
community is you have the Asterisk server on a public IP, then the
clients (soft or hard) can be behind NATs.  And all media (RTP) is run
through the Asterisk server (canreinvite=no), not directly to the peer. 
    Also configure the Asterisk server to ignore the IP given inside
the SIP packets and use the IP the SIP packet came from (nat=yes).

This is because most NAT gateways do not snoop into SIP packets and 
rewrite/get state of the RTP, to pre-open a NAT state entry, like
what is done for active FTP transfers.

So, in this case SIP is more of a client/server model, than peer2peer.

-- 
END OF LINE
       --MCP
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