[Linphone-users] RTP relaying

[Hans Georg Schaathun]

Dear all,

could someone help shed a little bit of light on the use of the 
SIP server as a relay for RTP (video/audio) traffic?

Using Linphone and sip.linphone.org, I observe that all traffic
is routed via the SIP server, even when both clients have public IP
addresses (no NAT) and should be able to route directly (once the
session is initiatied).

Is it the server or the clients who decide whether to use RTP
relaying?  I was not able to find any configuration option for it
in the android client, but I am not sure I have the right jargon
to find it.  Is it possible to change the setting?

If I understand it correctly, the RTP relaying is not covered by
SIP.  Is there an open standard to explain what is going on?
In particular, will the the server be able to route any video/audio
stream oblivious to the codec in use?

Thank you very much for your help.
-- 
:-- Hans Georg