Re: [Serusers] NATHELPER/rtpproxy does not work

wath version the SER do you use?

On Wed, 23 Feb 2005 15:15:27 -0500, Freddy - VoiceFinder SA freddy2006@gmail.com wrote:

Hello My SER implementation includes Asterisk voicemail for unavailable users, Radius Accounting, Digest Authentication and PSTN gateway forwarding, everything works very well but now I am trying NatHelper/rtpproxy for nated endpoints, nated clients are registering with public IP but I cannot hear incoming audio in nated X-lite clients even if I use Port Forwarding or enable DMZ in NAT device (LinkSys), I am very confused because I can hear audio from Asterisk... maybe I have some problem in ser, please take a look to my configuration file and send me some advice, thanks

rafael

PS: myser.cfg based on http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/nathelper.cfg?r...

/usr/local/etc/ser/Ser_VM_RadAcc_NatHelp-Test1.cfg

# Version: We are using: Sip EXpress router (0.8.99-dev1 (i386/linux)) (Agosto 2004) # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper

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

#/* Uncomment these lines to enter debugging mode debug=9 fork=yes log_stderror=yes #*/

listen=100.110.*.* listen=127.0.0.1 port=5060

# hostname matching an alias will satisfy the condition uri==myself". alias=my.domain.com.pe alias=100.110.*.*

check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/ser_fifo"

# sip_warning - Should replies include extensive warnings? # By default yes, it is good for trouble-shooting. sip_warning=yes

# ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/textops.so"

# digest authentication loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so"

# !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so"

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

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

# storing passwords in our database in plain text: # modparam("auth_db", "calculate_ha1", yes) # modparam("auth_db", "password_column", "password")

# For Rad Accounting modparam("acc", "radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc

modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 30 ) modparam("tm", "wt_timer", 20 )

# add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)

modparam("group", "db_url", "mysql://ser:heslo@localhost/ser") # "mysql" in cvs head version # modparam("uri", "db_url", "sql://ser:heslo@localhost/ser") # "sql" in ser0814 modparam("uri_db", "db_url", "mysql://ser:heslo@localhost/ser") # "mysql" in cvs head version

# ------------- registration parameters modparam("registrar", "nat_flag", 6) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 86400) modparam("registrar", "default_expires", 3600) modparam("registrar", "desc_time_order", 1) modparam("registrar", "append_branches", 1)

# !! Nathelper # modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT

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

route {

    log(1, "-------------------------------------------\n");
    log(1, "entering main loop\n");

    # 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");
            break;
    };
    if ( msg:len >= max_len ) {
            sl_send_reply("513", "Message too big");
            break;
    };

    # !! Nathelper
    # Special handling for NATed clients; first, NAT test is
    # executed: it looks for via!=received and RFC1918 addresses
    # in Contact (may fail if line-folding is used); also,
    # the received test should, if completed, should check all
    # vias for rpesence of received
    if (nat_uac_test("3")) {
            # Allow RR-ed requests, as these may indicate that
            # a NAT-enabled proxy takes care of it; unless it is
            # a REGISTER

            if (method == "REGISTER" || ! search("^Record-Route:")) {
                log("LOG: Someone trying to register from private

IP, rewriting\n");

                # This will work only for user agents that support symmetric
                # communication. We tested quite many of them and

majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling".

                fix_nated_contact(); # Rewrite contact with source

IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; };

    # 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");
            # t_relay(); ### use If don't want to enforce RTP proxy
            route(1);    ### Nathelper!!
            break;
    };

    # set Flag for Radius Accounting:

            if (method=="INVITE") {
            log(1, "INVITE MESSAGE RECEIVED - START ACC\n");
            setflag(1); /* set for accounting (the same value as

in log_flag!) */ };

            if (method=="BYE") {
            log (1, "BYE  - STOP ACCOUNTING\n");
            setflag(1);
            };

            if (method=="CANCEL") {
            log (1, "CANCEL - STOP ACCOUNTING\n");
            setflag(1);
            };

    setflag(3); # Set Radius Missed Flag (radius_missed_flag

param...)

    if (!uri==myself) {
            # mark routing logic in request
            append_hf("P-hint: outbound\r\n");
            # t_relay();
            route(1);
            break;
    };

    if (uri==myself) {

            if (method == "REGISTER") {
                    log(1, "ANALYZING REGISTER REQUEST\n");
                    # to use digest authentication
                    if (!www_authorize("my.domain.com.pe", "subscriber")) {
                            www_challenge("my.domain.com.pe", "0");
                            break;
                    };
                    if (!save("location")) {
                            sl_reply_error();
                    };
                    break;
            };

            /* ***************** very insecure Dial out to PSTN

logic ****************** */ ### Pendiente agregar seguridad a esta etapa, usar Digest-Auth o "credentials" ### ver http://www.voip-info.org/wiki-SER+example+pstn

            # forward n digit requests to gateway AS5350
            if(uri=~"^sip:9"){
                    log(1,"n digit expression match - Celulares Lima");
                    rewritehostport("100.110.*.*:5060");
                    route(2);
                    break;
            };

            # forward international calls to Asterisk (using Oh323

module to connect with H323 GWs) if(uri=~"^sip:00"){ rewritehostport("100.110.*.*:5060"); log(1,"n digit expression match - LDI"); route(2); break; };

            /*

---

*/

            lookup("aliases");
            if (!uri==myself) {
                    append_hf("P-hint: outbound alias\r\n");
                    # t_relay();
                    route(1);
                    break;
            };

            # does the user wish redirection on no availability?

(i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc

            if (is_user_in("Request-URI", "voicemail")) {
                    log(1, "requested user is in voicemail group");
                    setflag(4);
            };

            # native SIP destinations are handled using our USRLOC DB
            if (!lookup("location")) {
                    log(1,"unable to locate user");
                    # handle user which was not found
                    route(4);
                    break;
            };

    }; # End of "if(uri==myself)"

    append_hf("P-hint: usrloc applied\r\n");
    route(1);

    # if user is on-line and is in Voicemail group, enable redirection
    if (method == "INVITE" && isflagset(4)) {
            log(1, "invite for voicemail user->initiate failureroute[1]\n");
            t_on_failure("1");
    };

    # t_relay();

}

route[1] { # !! Nathelper if (uri=~"[@:](192.168.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; };

    # if client or server know to be behind a NAT, enable relay
    if (isflagset(6)) {
        force_rtp_proxy();
    };

    # NAT processing of replies; apply to all transactions (for example,
    # re-INVITEs from public to private UA are hard to identify as
    # NATed at the moment of request processing); look at replies
    t_on_reply("1");

    # send it out now; use stateful forwarding as it works reliably
    # even for UDP2TCP
    if (!t_relay()) {
            sl_reply_error();
    };

}

# !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

# ----------------- SIP-to-PSTN call routed -------------------

route[2]{ log(1,"route[2]:SIP-to-GW call routed"); if(!t_relay()){ sl_reply_error(); }; }

# --------------- Handling of Unavailable user ---------------- route[4] {

    # non-Voip -- just send "off-line"
    if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" ||

method=="BYE")) { sl_send_reply("404", "Not Found"); acc_rad_request("404 Not Found"); break; };

    # not voicemail subscriber
    if (!isflagset(4)) {
            sl_send_reply("404", "Not Found and no voicemail turned on");
            acc_rad_request("404 Not Found");
            break;
    };

  ### Forward to * voicemail adding prefix "vm" to simplify *

"extension.conf" to this: ### exten => _vmXXXXXXX,1,Voicemail(u${EXTEN:2}) ### exten => _vmXXXXXXX,2,Hangup

    prefix("vm");
    rewritehostport("100.110.**.**:5060");
    t_relay_to_udp("100.110.**.**","5060");

}

# if forwarding downstream did not succeed, try voicemail running at Asterisk

failure_route[1]{ if (t_check_status("485")){ revert_uri (); prefix("vm"); rewritehostport ("100.110.**.**:5060"); append_branch(); t_relay(); break; } }

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