[OpenSIPS-Users] Need help Nathelper + rtpproxy

[ha do]

Hi all

i set up rtpproxy run in same machine with opensips

my network topology:
ip phone1 (192.168.1.6) --------(192.168.1.248)opensips(172.26.0.2)-------(172.26.0.100)ip phone 2

media :
ip phone1 (192.168.1.6) --------(192.168.1.248)rtpproxy(172.26.0.2)-------(172.26.0.100)ip phone 2

i start rtpproxy : 
rtpproxy -l 172.26.0.2/192.168.1.248 -f -F -s udp:127.0.0.1:22222 -d DBUG:LOG_LOCAL7

the IP Phone 2 call IP Phone 1 and i did successfull on signaling + media 
when i disconnect the call i didnt see the command tear down the media session on rtpproxy
 
it is normal or i mis-config  the opensips.cfg, please help


Thank you
Ha

here is my opensips.cfg:

# ----------- global configuration parameters ------------------------

debug=9            # debug level (cmd line: -dddddddddd)

fork=yes

log_facility=LOG_LOCAL7

log_stderror=no    # (cmd line: -E)

children=4

port=5060



# ------------------ module loading ----------------------------------

#set module path

mpath="/usr/local/lib/opensips/modules/"

loadmodule "db_mysql.so"

loadmodule "signaling.so"

loadmodule "sl.so"

loadmodule "tm.so"

loadmodule "rr.so"

loadmodule "maxfwd.so"

loadmodule "usrloc.so"

loadmodule "registrar.so"

loadmodule "textops.so"

loadmodule "mi_fifo.so"

loadmodule "uri.so"

loadmodule "xlog.so"

loadmodule "nathelper.so"

#loadmodule "snmpstats.so"



# ----------------- setting module-specific parameters ---------------

# -- mi_fifo params --

modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")

# -- usrloc params --

#modparam("usrloc", "db_mode", 0)

# Uncomment this if you want to use SQL database

# for persistent storage and comment the previous line

modparam("usrloc", "db_url", "mysql://opensips:opensipsrw@localhost/opensips")

modparam("usrloc", "db_mode", 2)



# -- rr params --

# add value to ;lr param to make some broken UAs happy

modparam("rr", "enable_full_lr", 1)

modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:22222")

modparam("nathelper", "nortpproxy_str", "")

# -------------------------  request routing logic -------------------



# main routing logic

route{

        # initial sanity checks -- messages with

        # max_forwards==0, or excessively long requests

        if (!mf_process_maxfwd_header("10")) {

                sl_send_reply("483","Too Many Hops");

                exit;

        };



        if (msg:len >=  2048 ) {

                sl_send_reply("513", "Message too big");

                exit;

        };



        # we record-route all messages -- to make sure that

        # subsequent messages will go through our proxy; that's

        # particularly good if upstream and downstream entities

        # use different transport protocol

        if (!method=="REGISTER")

            record_route();

        # subsequent messages withing a dialog should take the

        # path determined by record-routing

        if (loose_route()) {

            # mark routing logic in request

            append_hf("P-hint: rr-enforced\r\n");

            route(1);

        };



        if (!uri==myself) {

            # mark routing logic in request

            append_hf("P-hint: outbound\r\n");

            route(1);

        };



        # if the request is for other domain use UsrLoc

        # (in case, it does not work, use the following command

        # with proper names and addresses in it)

        if (uri==myself) {

                if (method=="REGISTER") {

                    save("location");

                    exit;

                };

        }

        # native SIP destinations are handled using our USRLOC DB

        if(method=="INVITE"){

            if (dst_ip == 192.168.1.248)

                force_rtp_proxy("oei");

            if (dst_ip == 172.26.0.2)

                force_rtp_proxy("oie");

            t_on_reply("1");

        };

       if (is_method("BYE"))

                    unforce_rtp_proxy();

       

        if (!lookup("location","m")) {

            switch ($retcode) {

                case -1:

                case -3:

                    t_newtran();

                    t_on_failure("1");

                    t_reply("404", "Not Found");

                    exit;

                case -2:

                    sl_send_reply("405", "Method Not Allowed");

                    exit;

                }

            }

        route(1);

}

route[1] {

        # send it out now; use stateful forwarding as it works

        # reliably even for UDP2TCP

        failure_route[1];

        if (!t_relay()) {

                sl_reply_error();

        };

        exit;

}

onreply_route[1]{

    if (status=="200"){

        if(dst_ip == 172.26.0.2)

                force_rtp_proxy("oie");

        if(dst_ip == 192.168.1.248)

                force_rtp_proxy("oei");

    }

}



failure_route[1]{

        unforce_rtp_proxy();

}



when i make call and check on rtpproxy debug and see the rtpproxy debug :

DBUG:handle_command: received command "18781_4 UIEc0,18,4,97,9,2,15,8,101 09d614a45c92f2b0 at 172.26.0.100 172.26.0.100 2908 824bcd8bb5ba14fa;1"
INFO:handle_command: new session 09d614a45c92f2b0 at 172.26.0.100, tag 824bcd8bb5ba14fa;1 requested, type strong
INFO:handle_command: new session on a port 48190 created, tag 824bcd8bb5ba14fa;1
INFO:handle_command: pre-filling caller's address with 172.26.0.100:2908
DBUG:doreply: sending reply "18781_4 48190 192.168.1.248
"
DBUG:handle_command: received command "18780_4 LEIc0,101 09d614a45c92f2b0 at 172.26.0.100 192.168.1.6 17206 824bcd8bb5ba14fa;1 49ee0e488eccead5;1"
INFO:handle_command: lookup on ports 48190/42508, session timer restarted
INFO:handle_command: pre-filling callee's address with 192.168.1.6:17206
DBUG:doreply: sending reply "18780_4 42508 172.26.0.2
"
INFO:process_rtp: session timeout
INFO:remove_session: RTP stats: 238 in from callee, 323 in from caller, 561 relayed, 0 dropped
INFO:remove_session: RTCP stats: 1 in from callee, 0 in from caller, 1 relayed, 0 dropped
INFO:remove_session: session on ports 48190/42508 is cleaned up





      
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