Hi Daniel

Thank you so much for your feedback.

I came upon the below post, but I'm unable to find the example Norman posted.

http://www.mail-archive.com/users@openser.org/msg01932.html

In short the end solution is:

  # If both clients are behind nat we can check to see if they are behind the same

  # external IP and optimize the media path to go directly between them

else if ( isflagset(2) and isflagset(3) )

        {

            log(1, "Both Clients are behind NAT");

            # Store the destination domain into an AVP

            avp_printf("i:450", "$dd");

       

            if ( avp_check("i:450", "eq/$src_ip/g") )

            {

                xlog("L_INFO", "Detected Two Clients Behind the Same NAT - Disabling Mediaproxy");

           

                # Do not use mediaproxy as the clients seem to be behind the same NAT

                resetflag(2);

                resetflag(3);

            }           

       

        }

I haven't tested yet, but would I be right in saying my config should look something like:

####### Routing Logic ########


# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {

       # per request initial checks
       route(REQINIT);

       # NAT detection
       route(NAT);




# Caller NAT detection route
route[NAT] {
#!ifdef WITH_NAT
       force_rport();
       if (nat_uac_test("19")) {
               if (method=="REGISTER") {
                       fix_nated_register();
               } else {
                       fix_nated_contact();
               }
               setflag(FLT_NATS);

               # If both clients are behind nat we can check to see if they are behind the same

               # external IP and optimize the media path to go directly between them

               else if ( isflagset(FLT_NATS) and isflagset(FLT_NATB) ) {

                                log(1, "Both Clients are behind NAT");

               # Store the destination domain into an AVP

                                avp_printf("i:450", "$dd");

       

                                if ( avp_check("i:450", "eq/$src_ip/g") ) {

                                                         xlog("L_INFO", "Detected Two Clients Behind the Same NAT - Disabling RTPproxy");

           

                                # Do not use rtpproxy as the clients seem to be behind the same NAT

                                                         resetflag(FLT_NATS);

                                                         resetflag(FLT_NATB);

                                }

                }

         }

#!endif
       return;
}

# RTPProxy control
route[RTPPROXY] {
#!ifdef WITH_NAT
       if (is_method("BYE")) {
               unforce_rtp_proxy();
       } else if (is_method("INVITE")){
               force_rtp_proxy();
       }
       if (!has_totag()) add_rr_param(";nat=yes");
#!endif
       return;
}

Rgds

Deon

On Dec 20, 2010, at 2:41 PM, Daniel-Constantin Mierla wrote:

Hello,

On 12/20/10 11:25 AM, Deon Vermeulen wrote:

Hi List

Can someone please help me?

I need to implement PBX Services for a customer that has 10 IP Phones at his Main site and then 20 IP Phones located over 5 Branches.

My goal is to have RTP go end to end and not proxied.

I had a look at the below link and tried  it, but with no luck.

http://kamailio.org/dokuwiki/doku.php/examples:caller-callee-behind-same-nat

Here is a partial output of my kamailio.cfg file if someone would be so kind to help me.

#!ifdef WITH_NAT

# ----- rtpproxy params -----

modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722")

# ----- nathelper params -----

modparam("nathelper", "natping_interval", 30)

modparam("nathelper", "ping_nated_only", 1)

modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)

modparam("nathelper", "sipping_from", "sip:pinger@x.x.x.x")

# params needed for NAT traversal in other modules

modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")

modparam("usrloc", "nat_bflag", FLB_NATB)

#!endif

####### Routing Logic ########

# Main SIP request routing logic

# - processing of any incoming SIP request starts with this route

route {

        # per request initial checks

        route(REQINIT);

        # NAT detection

        route(NAT);

# Caller NAT detection route

route[NAT] {

#!ifdef WITH_NAT

        force_rport();

        if (nat_uac_test("19")) {

                if (method=="REGISTER") {

                        fix_nated_register();

                } else {

                        fix_nated_contact();

                }

                setflag(FLT_NATS);

        }

#!endif

        return;

}

# RTPProxy control

route[RTPPROXY] {

#!ifdef WITH_NAT

        if (is_method("BYE")) {

                unforce_rtp_proxy();

        } else if (is_method("INVITE")){

                force_rtp_proxy();

        }

        if (!has_totag()) add_rr_param(";nat=yes");

#!endif

        return;

}

what you have to do is to check if the source is natted and $si==$dd, then callee and caller are behind same nat (if there is one level of nat), so you don't force rtpproxy usage.

Cheers,
Daniel

--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com

-- 
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com