20 Jul
2005
20 Jul
'05
11:22 p.m.
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(a)yahoo.fr>
To: "Sebastian Kühner" <skuehner(a)veraza.com>
Sent: Wednesday, July 20, 2005 5:22 PM
Subject: Re: [Serusers] ACK
did you check /var/run/*/rtpproxy.pid
--- Sebastian Kühner <skuehner(a)veraza.com> a écrit :
___________________________________________________________________________
Hi!
Thanks for your question ;-)
I'm using Slackware...
----- Original Message -----
From: "harry gaillac" <gaillacharry(a)yahoo.fr>
To: "Sebastian Kühner" <skuehner(a)veraza.com>
Sent: Wednesday, July 20, 2005 5:07 PM
Subject: Re: [Serusers] ACK
=== message truncated ===
What's your distro Debian, .. ?
--- Sebastian Kühner <skuehner(a)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(a)yahoo.fr>
> To: "Sebastian Kühner" <skuehner(a)veraza.com>
> Sent: Wednesday, July 20, 2005 1:44 PM
> Subject: Re: [Serusers] ACK
>
>
> > your rtpproxy should work !
> >
> > --- Sebastian Kühner <skuehner(a)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(a)teigre.com>
> > > To: "Sebastian Kühner"
<skuehner(a)veraza.com>om>;
> > > <serusers(a)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(a)ohlmeier.org>
> > > > > To:
<serusers(a)lists.iptel.org>
> > > > > Cc: "Sebastian Kühner"
<skuehner(a)veraza.com>
> > > > > Sent: Tuesday, July 19,
2005 3:58 PM
Appel audio GRATUIT partout dans le monde avec le
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