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
> > > Subject: Re: [Serusers] ACK
> > >
> > >
> > > Hi,
> > >
> > > On Tuesday 19 July 2005 20:53, Sebastian
Kühner
> wrote:
> > >> I have two phones behind a Port Restricted
Cone
> NAT (both in the same
> > >> private area) and ser is running with
another
> public IP.
> > >>
> > >> I want to call from one of those phone to
the
> other. The call is set
> > >> up and I can talk, but one Softphone shows
me
> the message: "Waiting
> > >> acknowledgement..."... and all followed SIP
> messages don't reach the
> > >> other phone. I'm using a STUN server.
> > >>
> > >> Call from 14@xxx.xxx.xxx.xxx:5060 to
> 13@xxx.xxx.xxx.xxx:1024:
> > >>
> > >> 14 -> ser:
> > >> ----------
> > >> IVITE 13@ip.of.ser.xxx@5060 (Contact:
> 14@192.168.1.101:5060)
> > >>
> > >> ser -> 13:
> > >> ----------
> > >> INVITE 13@xxx.xxx.xxx.xxx:1024 (Contact:
> 14@xxx.xxx.xxx.xxx:5060)
> > >
> > > sorry but what do you use STUN for if the
UAs
> still use their private
> > > IPs and
> > > your SER is re-writting the Contact? If you
> allready fixing the IP it