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@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 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@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 ===
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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers