Using a media relay is the solution for your problem. You are asking for a solution to not use the solution which makes no sense.
Adrian On Feb 25, 2013, at 7:24 PM, Roberto Spadim wrote: > humm i got the same problem but didn't found a solution > my solution was connect internet (public ip) directly to voip server, in > other words, i removed the opensip proxy and ntpproxy, but if anyone have the > solution could be very nice, i googled many examples but they don't work > > > 2013/2/25 Muhammad Shahzad <shaherya...@gmail.com> > You are missing one fundamental fact, that is you have to handle NAT for both > signalling and media. From your description it looks signalling is going > perfect (NAT is correctly handled), since you are able to establish call > between two clients successfully, clients can register, make call, accept > call and hangup call with your server. So main goal of NAT Traversal module > is achieved. > > However, there is no media on call, so media NAT is not handled. NAT > Traversal and / or NAT Helper modules may try to fix media NAT issues as well > by manipulating SDP but in so many case they will be simply NOT enough for > this purpose. Especially in case of 3g and corporate networks, which may have > very very complex network typology with multiple layers of NAT (so called > Nested NAT). So rtp / media proxy is the ONLY solution that can handle media > across such complex networks. > > If you have really good sip clients with support for STUN / TURN / ICE etc. > and you somewhat control over client data network environment, them you may > fix media NAT issues up to 90% but in about 5-10% cases you will still need a > media relay. > > Thank you. > > > On Mon, Feb 25, 2013 at 11:51 PM, leo <uzcud...@yahoo.it> wrote: > Hello, > > Unfortunately after reading the forum i've to open a new post about NAT > because i couldn't find a clear solution and information for my problem. > I've also read the NAT Traversal module documentation. > > I've an OpenSIPS server (version 1.8.2) on a Debian system (6.0.7 - > 2.6.32-5-686). > OpenSIPS was installed by the apt-get install using the apt.opensips.org > repository and configured with osipsconfig (residential script with ALIASES, > AUTH, DBACC, DBUSRLOC and DIALOG). > > The UAs can register to the OpenSIPS server. They can place the call but i > 've no audio no video. > The OpenSIPS server has a public IP address (so, no natted). > The UAs could be natted or with public ip thru 3G. > > I wouldn't like to use rtproxy or mediaproxy cause the rtp traffic would be > passing by those servers (am i correct?) adding jitter and latency. > I would set up the system in the way the the rtp traffic would be P2P. Would > NAT Traversal be the solution? How it should be configured (i've already > enabled the required modules too)? > > Thanks a lot. > > Leo. > > > > -- > View this message in context: > http://opensips-open-sip-server.1449251.n2.nabble.com/NAT-tp7584918.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_78...@hotmail.com > Email: shaherya...@googlemail.com > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Roberto Spadim > SPAEmpresarial > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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