Re: [SR-Users] SIP UA's behind same NAT

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