New subject: [Serusers] ACK

20 Jul 2005


      Hi,
I don't have a rtpproxy.pid file. You mean the *.sock file?
Here is the permission:
srwxr-xr-x  1 root  root     0 2005-07-20 18:18 rtpproxy.sock=
The rtpproxy has to create a pid-file?
Thanks!
----- Original Message ----- 
From: "harry gaillac" gaillacharry@yahoo.fr
To: "Sebastian Kühner" skuehner@veraza.com
Sent: Wednesday, July 20, 2005 5:22 PM
Subject: Re: [Serusers] ACK
...
did you check /var/run/*/rtpproxy.pid
--- Sebastian Kühner skuehner@veraza.com a écrit :
...
Hi!
Thanks for your question ;-)
I'm using Slackware...
----- Original Message ----- 
From: "harry gaillac" gaillacharry@yahoo.fr
To: "Sebastian Kühner" skuehner@veraza.com
Sent: Wednesday, July 20, 2005 5:07 PM
Subject: Re: [Serusers] ACK
...
What's your distro Debian, .. ?
--- Sebastian Kühner skuehner@veraza.com a écrit
:
...
...
It should... but it doesn't. I have ser 0.9.0
and
...
...
the latest rtpproxy
version.
WARNING: rtpp_test: can't get version of the RTP
proxy
----- Original Message ----- 
From: "harry gaillac" gaillacharry@yahoo.fr
To: "Sebastian Kühner" skuehner@veraza.com
Sent: Wednesday, July 20, 2005 1:44 PM
Subject: Re: [Serusers] ACK
...
your rtpproxy should work !
--- Sebastian Kühner skuehner@veraza.com a
écrit
...
...
:
...
...
Hi,
Ok, my rtpproxy doesn't work, so I try it
with
...
...
STUN.
...
...
When I look at my
SIP-messages I get the information, that the
audio
...
...
stream has to go through
my public IP... but I don't hear anything (I
have
...
...
the volume on maximum).
The Invite comes with this message:
v=0.
o=- 3330865830 3330865830 IN IP4
xxx.xxx.xxx.xxx.
...
...
 <-- Public IP

s=SJphone.
c=IN IP4 xxx.xxx.xxx.xxx
<--
...
...
...
...
Public IP
t=0 0.
a=direction:active.
m=audio 16482 RTP/AVP 3 8 0 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11,16.
Doesn't that mean, that the audio-stream has
to
...
...
go
...
...
through my public IP now?
Both sides doesn't hear anything...
What's wrong?
Sebastian
----- Original Message ----- 
From: "Greger V. Teigre" greger@teigre.com
To: "Sebastian Kühner"
skuehner@veraza.com;
...
...
...
...
serusers@lists.iptel.org
Sent: Wednesday, July 20, 2005 2:24 AM
Subject: Re: [Serusers] ACK
> Sebastian,
> I know many people don't like STUN.
However, I
...
...
...
...
have good experiences with
> STUN and prefer to use STUN as a "first
layer
...
...
...
...
defence."  For many NATs I
> then avoid the proxying. However, there
are
...
...
some
...
...
things that can go wrong:
> For one, you need to make sure that the
STUN
...
...
...
...
server is running correctly
on
> two ports and two IP addresses. If you for
example
...
...
have a firewall
blocking
> one port, STUN will give the wrong result.
But
...
...
the
...
...
biggest problem can be
> faulty STUN implementations in the EUCs.
They
...
...
...
...
normally behave ok for the
> most standard NATs, but there are some
non-standard NATs and the EUC's
> behavior can be unpredictable.  Also, some
EUCs
...
...
try to rewrite the IP:port
> even if they are behind a symmetric NAT
(or if
...
...
the
...
...
STUN server is not
> correctly set up, the EUC will conclude
with
...
...
the
...
...
wrong result).
>     If you know the clients you are going
to
...
...
use,
...
...
you can test and limit
the
> problems and STUN can be a great cost
saver!
...
...
If
...
...
your gateway supports
> active media (direction=active), then you
only
...
...
...
...
have IP-2-IP phone calls to
> proxy.
>
> To your question: Sipura has a good
implementation
...
...
of STUN, but has MANY
> options for NAT. Your problem is that the
RTP
...
...
and
...
...
RTCP is not traversing
the
> NAT to your Sipura.  Either you don't
force
...
...
...
...
proxying in onreply for OKs,
or
> something goes wrong.  An ngrep trace of
the
...
...
call
...
...
setup will reveal what
the
> problem can be.
> g-)
>
> Sebastian Kühner wrote:
> > Thank you Nils,
> >
> > Now it's working better!
> >
> > The problem that I have now is that I
don't
...
...
hear
...
...
anything if I call
> > from the SIPURA to a Gateway, but the
callee
...
...
is
...
...
hearing me.
> >
> > What could be the problem of that
one-way
...
...
...
...
conversation? Had anyone of
> > you the same problem using a Restricted
Cone
...
...
...
...
NAT?
> >
> > Thanks!
> >
> > Sebastian
> >
> >
> > ----- Original Message -----
> > From: "Nils Ohlmeier"
lists@ohlmeier.org
...
...
...
...
> > To: serusers@lists.iptel.org
> > Cc: "Sebastian Kühner"
skuehner@veraza.com
...
...
...
...
> > Sent: Tuesday, July 19, 2005 3:58 PM
=== message truncated ===
___________________________________________________________________________
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