[Telepathy] telepathy-sofiasip parameters
alsuren at gmail.com
Mon Jan 31 05:30:31 PST 2011
> I'm working on a telepathy-accounts-kcm plugin for telepathy-sofiasip and
> the in-code documentation of the provided configuration parameters are not
> sufficient for me to be able to produce a sane configuration UI. Therefore
> some questions concerning telepathy-sofiasip's parameters:
> discover-stun, stun-server, stun-port:
> In the code i found that discover-stun is ignored if stun-server is set,
> therefore if a user sets discover-stun to TRUE stun-server needs to be
> empty to make this configuration work?
I didn't know this is how it worked. Just have a ticky-box for discover, and
otherwise let the user fill in stun server and port if this is how it seems to
work? (it might be sensible to change the stun server configuration to be the
same as gabble's once we do ICE)
> local-ip-address / local-port:
> what are these used for, are they overriden by anything, does
> discover-binding affect this?
I think this just changes some headers in SIP packets. Most sip servers that
understand NATs shouldn't need this to be changed.
> proxy-host / port / transport:
> does proxy-host need to have the format "sip:<FQDN/IP>"?
> is transport really related to the proxy?
for people behind firewalls, you can have a setup like:
sofiasip <-> outbound proxy <-+-> registrar (login server)
| +-> other SIP server <-> other user
> extra-auth-user / extra-auth-password:
> what are extra authentication challenges?
in the above diagram, "other SIP server" may also require authentication.
Extra auth user/pass are used for this. The primary auth credentials are used
if these don't exist. We are currently limited to one extra set of sip
credentials. SIP authentication challenges are actually SASL, so we could in
theory do away with all this nonsense and pop up a SASL channel whenever
someone asks us to authenticate.
> 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.
> Florian Reinhard
> telepathy mailing list
> telepathy at lists.freedesktop.org
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