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 <[email protected]> > 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 <[email protected]> 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 >> [email protected] >> 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: [email protected] > Email: [email protected] > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Roberto Spadim SPAEmpresarial
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