[Serusers] ACK

Sebastian Kühner skuehner at veraza.com
Thu Jul 21 14:53:48 CEST 2005


Hi!

I tried to use the "modparam("nathelper", "rtpproxy_sock",
"udp:10.56.0.5:22222")"... but without result.

I downloaded the ONsip-package (without using it... postgres is implemented
there?), but I didn't find the Getting Started guide...



----- Original Message ----- 
From: "Greger V. Teigre" <greger at teigre.com>
To: "Sebastian Kühner" <skuehner at veraza.com>; <serusers at lists.iptel.org>
Sent: Thursday, July 21, 2005 2:20 AM
Subject: Re: [Serusers] ACK


> Just curious: Have you used the ONsip.org Getting Started guide, configs
and
> getting started source package!?  Rtpproxy is included, while mediaproxy
is
> a standalone package where everything is prepared.
>
> Getting this far, you can try to the udp mode:
> # We set up requests over udp
> modparam("nathelper", "rtpproxy_sock", "udp:10.56.0.5:22222")
>
> Start rtpproxy this way:
> rtpproxy -l your_up -s udp:*
>
> g-)
>
>
> Sebastian Kühner wrote:
> > 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 at yahoo.fr>
> > To: "Sebastian Kühner" <skuehner at veraza.com>
> > Sent: Wednesday, July 20, 2005 5:22 PM
> > Subject: Re: [Serusers] ACK
> >
> >
> >> did you check /var/run/*/rtpproxy.pid
> >>
> >> --- Sebastian Kühner <skuehner at veraza.com> a écrit :
> >>
> >>> 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
> >>>
> >> === message truncated ===
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >
___________________________________________________________________________
> >> Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo!
> >> Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com
> >>
> >>
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
>





More information about the sr-users mailing list