[Serusers] Phone rings but no audio - sr-users

31 Jan 2005

Hello,

I'm running ser 0.8.14 and rtpproxy CVS on debian sarge.
You can find my ser.cfg. below.

I'm forwarding my calls to a Cisco AS5350 whish serves as a PSTN
gateway. I can place a call to a PSTN number and the phne rings, but
then no audio is sent or received. I use kphone as a client behind NAT.

thnaks

Luca

--- ser.cfg routing ---

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

        # 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
        record_route();
        # loose-route processing
        if (loose_route()) {
                t_relay();
                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
        };

        # 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);
                break;
        };

        if (!uri==myself) {
                # mark routing logic in request
                append_hf("P-hint: outbound\r\n");
                rewritehostport("$AS5350_IP:5060");
                route(1);
                break;
        };

        # 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") {

                        # Uncomment this if you want to use digest
authentication
                        if (!www_authorize("my.domain",
"subscriber")) {

                                www_challenge("my.domain", "0");
                                break;
                        };

                        save("location");
                        break;
                };

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

                # native SIP destinations are handled using our USRLOC
DB
                if (!lookup("location")) {
                        sl_send_reply("404", "Not Found");
                        break;
                };
        };
        append_hf("P-hint: usrloc applied\r\n");
        route(1);
}

# -- nathelper --
route[1]
{

        if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&& !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();
        };

}
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();
        };
}