Re: [Serusers] Phone rings but no audio

look in logfiles for debug for proxies, and see what they are doing

Iqbal

Luca Corti wrote:

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(); }; }

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