# RTP problem? **URL:** \ **Category:** Asterisk Support\ **Created:** [August 26, 2008, 2:19pm UTC](https://community.asterisk.org/t/rtp-problem/22726 "2008-08-26T14:19:28Z")\ **Posts on this page:** 2\ **Page:** 1
**Author:** ![steve-toto](https://avatars.discourse-cdn.com/v4/letter/s/dec6dc/32.png) [@steve-toto](https://community.asterisk.org/u/steve-toto)\ **Post date:** [August 26, 2008, 2:19pm UTC](https://community.asterisk.org/t/rtp-problem/22726/1 "2008-08-26T14:19:28Z")
Hi, I am running Asterisk 1.4.21 behind a NAT I have setup my asterisk server and my phones located intothe building are working fine. I have a VoIP provider that I am using -\> fine ! I can make phone called and receive without any problems. Now, I have ONE phone located outside the building. I can register If I dial the extension -\> the phone is ringing This phone can make ring my local phone, But : - I can’t hear anything - The person over there can hear me I thought about an NAT and RTP problems, but I really don’t know anymore what to do ?! ? I place a another phone, registering to my asterisk server via Public IP address, thinking that I will force the COM to get out from the router and come back. Not sure about this is really happen like this or not I forced it into the SIP file with CANIVITE=no CANREINVITE=no NAT=yes And this phone INSIDE the building works fine the other one OUTSIDE the building doesn’t work ( And both have the some SIP configuration [ CANINVITE=no, NAT=yes…etc,) I am lost … Any Idea ? Thanks and regards, ---
**Author:** ![steve-toto](https://avatars.discourse-cdn.com/v4/letter/s/dec6dc/32.png) [@steve-toto](https://community.asterisk.org/u/steve-toto)\ **Post date:** [August 27, 2008, 9:10am UTC](https://community.asterisk.org/t/rtp-problem/22726/2 "2008-08-27T09:10:44Z")
Found the solution… all I setup was : NAT=yes caninvite Canreinvite=… =\> This is only for client behind a NAT * * * My Asterisk Server is also behind a NAT =\> Added into SIP.CONF externip= Localnet=… srvlookups=yes