[Serusers] ACK

[Sebastian Kühner]

Hi!

Thanks for your question ;-)

I'm using Slackware...

----- Original Message ----- 
From: "harry gaillac" <gaillacharry at yahoo.fr>
To: "Sebastian Kühner" <skuehner at veraza.com>
Sent: Wednesday, July 20, 2005 5:07 PM
Subject: Re: [Serusers] ACK


> What's your distro Debian, .. ?
>
> --- Sebastian Kühner <skuehner at 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 at yahoo.fr>
> > To: "Sebastian Kühner" <skuehner at veraza.com>
> > Sent: Wednesday, July 20, 2005 1:44 PM
> > Subject: Re: [Serusers] ACK
> >
> >
> > > your rtpproxy should work !
> > >
> > > --- Sebastian Kühner <skuehner at 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 at teigre.com>
> > > > To: "Sebastian Kühner" <skuehner at veraza.com>;
> > > > <serusers at 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 at ohlmeier.org>
> > > > > > To: <serusers at lists.iptel.org>
> > > > > > Cc: "Sebastian Kühner" <skuehner at 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 at xxx.xxx.xxx.xxx:5060 to
> > > > 13 at xxx.xxx.xxx.xxx:1024:
> > > > > >>
> > > > > >> 14 -> ser:
> > > > > >> ----------
> > > > > >> IVITE 13 at ip.of.ser.xxx@5060  (Contact:
> > > > 14 at 192.168.1.101:5060)
> > > > > >>
> > > > > >> ser -> 13:
> > > > > >> ----------
> > > > > >> INVITE 13 at xxx.xxx.xxx.xxx:1024 (Contact:
> > > > 14 at 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
> >
> === message truncated ===
>
>
>
>
>
>
>
>
___________________________________________________________________________
> Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
> Téléchargez cette version sur http://fr.messenger.yahoo.com
>