22 Feb
2005
22 Feb
'05
1:06 a.m.
Java Rockx wrote:
I full agree with you. I see that we've taken
different approachs to
NAT. We do not use STUN because:
* the two opensource STUN projects are really not written well,no
offense to the authors :)
That is interesting. We use vovida stun and have not experienced any
problems at all. What exactly are you referring to?
* it adds a new layer of complexity to support
That's true.
* it forces the SIP users to have additional settings
on their SIP
phones/UAs
Ditto.
Instead we do 100% transparent RTP proxying with
mediaproxy. However,
we did not want all that RTP traffic to pass through our servers so we
__only__ proxy RTP streams when one or both SIP clients are behind a
NAT __or__ when a client is accessing the voicemail system (see my
other posts on how I have Asterisk on a 10.x IP network)
The benefit of doing NAT traversal this way is that we believe it
scales very well and is the most simple to maintain and support. We've
always taken the approach that we're going to put a million customers
on our VoIP platform and RTP is a show stopper if you can't keep up
with load demands.
We avoid this by making sure customers connect their IAD up as
follows:
INTERNET ----- Broadband Router ----- IAD ----- PC
We can do this because we only support IADs that are routers, such as
the UTstarcom iAN-02EX or the Grandstream 486. We also ship the IADs
locked down so that customers cannot modify their settings. The
UTstarcom allows us to remotely change the configuration. All this
translates to users having their IADs with public IPs and therefore,
RTP doesn't hit our servers.
Exactly! And that's the problem. If you are to support (like us), regular
SIP phones hooked up on a router somewhere, you cannot do this. Either you
proxy everything (which as you say, is very bad, indeed) or you use STUN...
But hey, we have now started to detail the reference design. What kind of
UAs should we support? ;-)
g-)