[Telepathy] telepathy-sofiasip parameters

[Kai.Vehmanen at nokia.com]

Hi,

On 31 Jan 2011, David Laban wrote:
>> since i do not exactly know how sip NAT traversal works, i guess that
>> stun, proxy and discover-binding (which seems to make sofiasip use natify
>> and rport) exclude another?
>> Are Natify and rport being used if there is a STUN server?
>
> Not *completely* sure either. I think that the discover-binding stuff
> doesn't actually affect NAT traversal for VoIP calls because the information
> doesn't get communicated to farsight. Again: I am thinking of ripping up some
> of this code when we port to Call and start using ICE, so you can probably hide
> most of these options.

NAT traversal for media (RTP) and signaling (SIP) are two different
things, so you cannot really get rid of anything on signaling side
when you start using ICE (which is for media traversal).

On signaling side, the official solution to this problem is sip-outbound,
but sofia-sip does not yet fully support it (albeit partial support
is already present and has been for a while):
http://tools.ietf.org/html/rfc5626

Both STUN and binding-discovery are basicly ways to do signaling NAT 
traversal with proxies that do not provide any support for client NAT 
traversal (they assume client can provide a publically routable IP:port 
contact on its own).

In real-life, there are plenty of proxies that fall in between: they
don't support full outbound yet, but they still help clients with 
NAT traversal (most important bit is that they ignore the SIP contact 
field and instead send signaling to the same port client's REGISTER came
from). Even with these proxies, the keepalive options of telepathy-sofiasip
might still be useful.