24 May
2007
24 May
'07
1:06 p.m.
John Breen wrote:
SIP wrote:
127.0.0.1 is a special address. If MY machine, for instance, tries to
connect to 127.0.0.1, it's connecting to itself. If Bob's machine tries
to connect to 127.0.0.1, he's connecting to Bob's machine. While,
technically, you COULD accomplish having Asterisk on an internal IP
address on the local machine, in order for it to actually work, you'd
need to turn the computer into a router (start doing RIP or some such),
which would immeasurably complicate things.
Remember, a SIP call has two parts. One is the signaling portion (the
actual SIP portion of the call) and one is the media portion (RTP -- the
voice stuff). SER handles the SIP portion but doesn't do anything with
the RTP portion. It can't, itself, 'proxy' that voice portion from the
outside world to an internal-only IP on your box, so you won't ever get
any audio to your Asterisk server that way. What happens in that
scenario is that your softphone calls the SER box. SER tells the
softphone that it needs to connect its RTP/audio portion of the call to
the IP of the asterisk server directly. If the IP of the asterisk server
is internal (or the loopback IP), then the softphone has no way to know
how to get to it, so it can't connect the audio properly.
If you were running mediaproxy or RTPproxy on the SAME box, IT could
proxy the audio portion from the outside world to the Asterisk server
running on the same machine with an internal IP. However, then you're
creating an incredibly complex scenario and using even more resources
for each call.
Your best bet if you need both servers to run on the same box is simply
to run Asterisk on the same external IP as SER but with a different
port. That way, the outside world can connect its audio to the Asterisk
server and you won't get some of the weird issues you're getting now.
N.
Your biggest issue, bar none, is using the
loopback address for
anything externally-facing. 127.0.0.1 is called a loopback address
for a reason -- it never leaves the host itself. Just how do you
expect the SIP phones to be able to contact a 127.0.0.1 address when
that address is, in essence, themselves?
There's something inherently flawed in your design.
Let's back up and explain, in detail, what it is exactly you're
trying to do, and why you feel the 127.0.0.1 address should be part
of this equation? Perhaps we can suggest something if we know what
you're trying to do.
There's a very simple reason. Asterisk and SER are on the same
machine. So the traffic from SER to asterisk doesn't need to leave
the box. This is why I'm using the loopback interface - there's no
need to use the real world one.