I am running asterisk behind nat, and 2 sip phones on 2 different adsl neted connections, asterisk is staying always in rtp media path, while canreinvite=yes is configured in both extensions. I need asterisk to stay away from the rtp media path, what is wrong with that setup? Regards, Osama Kamal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060604/b83a4415/attachment.htm
----- Osama Kamal <okamalo@gmail.com> wrote:> I am running asterisk behind nat, and 2 sip phones on 2 different adsl > neted connections, asterisk is staying always in rtp media path, while > canreinvite=yes is configured in both extensions. I need asterisk to > stay away from the rtp media path, what is wrong with that setup?It is nearly impossible to get a direct media path between two endpoints that are both behind NATs, regardless of the SIP server/proxy you use. Asterisk is no different in this regard. -- Kevin P. Fleming Senior Software Engineer Digium, Inc.
does thia apply on SIP only or also IAX? On 6/5/06, Kevin P. Fleming <kpfleming@digium.com> wrote:> > > ----- Osama Kamal <okamalo@gmail.com> wrote: > > I am running asterisk behind nat, and 2 sip phones on 2 different adsl > > neted connections, asterisk is staying always in rtp media path, while > > canreinvite=yes is configured in both extensions. I need asterisk to > > stay away from the rtp media path, what is wrong with that setup? > > It is nearly impossible to get a direct media path between two endpoints > that are both behind NATs, regardless of the SIP server/proxy you use. > Asterisk is no different in this regard. > > -- > Kevin P. Fleming > Senior Software Engineer > Digium, Inc. > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060605/cfde1aed/attachment.htm
----- Osama Kamal <okamalo@gmail.com> wrote:> does thia apply on SIP only or also IAX?It has nothing to do with the VOIP protocol, it is the nature of how NAT and port translation works. -- Kevin P. Fleming Senior Software Engineer Digium, Inc.