On 02/26/2009 01:19 PM, carl Lougher wrote: > Thanks for that. So does it mean by using rtpproxy you will therefore carry > all the rtp streams through that server yes, that is the role of RTPProxy - to proxy the RTP streams, therefore those go via the server.
If you want end-to-end RTP stream, then look at STUN, if the phones are not behind symmetric nat, it can help. > or can it be redirected to the sip provider from the endpoint? > > Also how do you put the kamailio server in the equation? Do you set it up as > an external proxy for the clients or do you register the clients to it then > just use asterisk for the media/vmail etc? > I do everything in kamailio but the media services which i do with asterisk (vmail, ivr, ...) - authentication, registration, call routing is done in kamailio. Cheers, Daniel > > --- On Thu, 26/2/09, Daniel-Constantin Mierla <mico...@gmail.com> wrote: > > >> From: Daniel-Constantin Mierla <mico...@gmail.com> >> Subject: Re: [Kamailio-Users] Kamailio Newb questions >> To: c_loug...@yahoo.co.uk >> Cc: users@lists.kamailio.org >> Date: Thursday, 26 February, 2009, 9:16 AM >> Hello, >> >> On 02/26/2009 12:59 AM, carl Lougher wrote: >> >>> Howdy, >>> I'm trying to remove the media/rtp streams from an >>> >> asterisk server for natted users so would like to know if >> this is possible with kamailio. >> >>> >>> >> yes it is possible. nathelper+rtpproxy is the option I use >> and prefer >> because of flexibility and performances. You can see an >> example at: >> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy >> >> >> >>> Qu's: >>> What is the best option? >>> rtpproxy/mediaproxy? >>> nathelper? >>> >>> If i use kamailio to achieve this does it mean that i >>> >> still have to carry the rtp streams through the kamailio >> server instead? >> >>> >>> >> through the rtpproxy server, which can be located on same >> or different >> machine than kamailio. >> >> >>> Also will i need to change the logon info for the >>> >> clients so they now logon to kamailio then i just point >> registrar to asterisk? >> >>> Can i use kamailio for sip trunks to asterisk and >>> >> carry rtp and natted clients media streams rather than >> register to asterisk? >> >>> >>> >> Yes, you can register to kamailio, see registrar and usrloc >> modules. >> >> Cheers, >> Daniel >> >> >>> Many thanks, >>> Taff.. >>> >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users@lists.kamailio.org >>> >>> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >>> >>> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> > > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users@lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
