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 IPs=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"
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"
> > To: serusers@lists.iptel.org > > Cc: "Sebastian Kühner"
> > Sent: Tuesday, July 19, 2005 3:58 PM
=== message truncated ===
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