You need to make sure your firewall in front of sipx has: 1. SPI disabked 2. SIP ALG disabled 3. Be performing (full cone) symmetric port NAT. 4. NAT'ing 5060/tcp&udp, 5080/udp and 30000-31000/udp and 5. Remote workers also need to be doing 1&2 on their firewall AND be set to be sending their internal (local) ip address (see registrations).
If this was all done it would be working. http://www.myitdepartment.net/support/Three_things_I_really_like_about_sipXecs_4.pdf see page 4 On Wed, Dec 1, 2010 at 7:03 PM, Arman Güngör <[email protected]> wrote: > Hi all, > > I have a sipX deployed behind NAT. I configured it according to > http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal. > > Now there are three phones, one of them is behind the same NAT with sipX. > The other two phones are connected to internet through different ADSLs, both > of them are behind NAT. > > When the call is established between the phone in local NAT and remote > user, RTPs are routed through sipX relay. But when the call is established > between remote users( NAT behind ADSLs ) RTPs are not routed through media > relay, so there is no audio on both sides. Why it does not route RTPs > through media relay? What should I do in order to enable media relay in this > case? > > Regards, > > Arman > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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