[OpenSER-Users] Openser + Mediaproxy not working with 2 domains

Mario Bedialauneta mbedial at gmail.com
Fri Jan 18 12:36:25 CET 2008


Hi all,
I've already solved the scenario 1. The only think to do was to comment the
lines

#if (!uri==myself) {

# mark routing logic in request

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

#route(1);

#};
and handle other uris in the way we do wit our uri.

However I'm not able to make the scenario 2 works. onreply_route[1] should
receives 200 OK with SDP and force to use the mediaproxy but it doesn't...
onreply_route[1] {

if (status=~"(183)|(2[0-9][0-9])") {

if (client_nat_test("1")) {

fix_contact();

};

use_media_proxy();

};

}
Thanks in advance for your help and best regards.



2008/1/17, Andreti <mbedial at gmail.com>:
>
>
> Hi everybody,
> I'm working with Openser + Mediaproxy 1.9.0 and it seems that everything
> is
> working when the calls are establised between users attached to the same
> proxy server, even with different kind of NATs.
>
> However It doen't work in 2 different scenarios, and the result is exactly
> the same , the video and audio is only sent in one way.
>
> Scenario 1
> ========
> User A attached to the SIP proxy xxx.xxx.xxx.13 (Public IP)  calls to a GW
> xxx.xxx.xxx.11 (Public IP)  with several users  internally associated. In
> this case the user A can see the video and audio sent by the GW,  but the
> GW
> doesn't receive any RTSP stream. It seems that the mediaproxy doesn't do
> anything, why? maybe because the GW blongs to other domain (xxx.xxx.xxx.11
> )
> ? What can I do?
> If the GW calls to user A, it works fine (I can see the session in the
> mediaproxy with sessions.py)
>
>
> Scenario 2
> ========
> In this case, I have another GW with Public IP address xxx.xxx.xxx.14, but
> it doesn't include in the INVITE message the SDP body. The GW calls to the
> same user attached to the SIP proxy xxx.xxx.xxx.13 , and the behaviour is
> exactly the same as scenario 1, the calling site can sse the video and
> audio
> but the called can't.
> Unlike the previous scenario, the signalling is:
>
> INVITE without SDP  -->  200 OK (SDP)  -- > ACK (SDP)
>
> In theory, Mediaproxy 1.9.0  should support this procedure since it's a
> SIP
> standard mechanism, however the called party doesn't receive RTP stream.
> In
> my opinion, the problem could be related to scenario 1, I mean , the
> calling
> party is not attached to the SIP proxy (belongs to other domain) and when
> the 200 OK (SDP) message arrives to the SIP proxy, the mediaproxy doesn't
> do
> anything
>
> Sorry for the complex explanation. I've waste a lot of time trying  to
> solve
> this solution and honestly I don't know what to do. Please, could somebody
> help??
>
> I attach my openser.conf. I hope it helps.
>
> Andreti
>
>
>
> debug=5         # debug level (cmd line: -dddddddddd)
> #fork=yes
> #log_stderror=no        # (cmd line: -E)
> log_facility=LOG_LOCAL0
>
> # Uncomment these lines to enter debugging mode
> #fork=no
>
> log_stderror=yes
>
> listen=xxx.xxx.10.12
>
> check_via=no    # (cmd. line: -v)
> dns=no           # (cmd. line: -r)
> rev_dns=no      # (cmd. line: -R)
> #port=5060
> #children=4
> fifo="/tmp/openser_fifo"
> fifo_db_url="mysql://openser:openserrw@localhost/openser"
>
> # ------------------ module loading ----------------------------------
> mpath = "/usr/local/lib/openser/modules/"
>
> # Uncomment this if you want to use SQL database
> loadmodule "mysql.so"
> loadmodule "domain.so"
> loadmodule "mediaproxy.so"
> loadmodule "uri_db.so"
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "rr.so"
> loadmodule "maxfwd.so"
> loadmodule "usrloc.so"
> loadmodule "registrar.so"
> loadmodule "textops.so"
> loadmodule "avpops.so"
> loadmodule "uri.so"
>
> loadmodule "xlog.so"
> loadmodule "acc.so"
> loadmodule "auth_radius.so"
> loadmodule "group_radius.so"
> loadmodule "avp_radius.so"
> loadmodule "nathelper.so"
>
>
> # Uncomment this if you want digest authentication
> # mysql.so must be loaded !
>
> loadmodule "auth.so"
> loadmodule "auth_db.so"
>
> # ----------------- setting module-specific parameters ---------------
> # -- 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_mode", 2)
> modparam("usrloc", "timer_interval", 20)
>
> # -- auth params --
> # Uncomment if you are using auth module
> #
>
> modparam("auth_db", "calculate_ha1", yes)
> #
> # If you set "calculate_ha1" parameter to yes (which true in this config),
> # uncomment also the following parameter)
> #
>
> modparam("auth_db", "password_column", "password")
>
> # -- acc params --
> modparam("acc", "radius_flag", 1)
> modparam("acc", "radius_missed_flag", 1)
> #cambio 16_04_07 modparam("acc", "radius_missed_flag", 2)
> modparam("acc", "log_flag", 1)
> modparam("acc", "log_missed_flag", 1)
> modparam("acc", "service_type", 15)
> modparam("acc|auth_radius|group_radius|avp_radius", "radius_config",
>    "/usr/local/etc/radiusclient-ng/radiusclient.conf")
> #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp")
> #puesto para el CDRTool
> modparam("acc", "failed_transaction_flag", 1)
> modparam("acc", "report_cancels",     0)
> modparam("acc", "report_ack",         0)
> modparam("acc", "early_media",        0)
> modparam("acc", "log_level",          1)
> #modparam("acc", "radius_config",      "/etc/openser/radius/client.conf")
> modparam("acc", "radius_extra",       "Sip-Src-IP=$si;Sip-Src-Port=$sp;\
>                                       Sip-RPid=$avp(s:rpid); \
>                                       Source-IP=$si; \
>                                       Source-Port=$sp; \
>                                       Canonical-URI=$avp($can_uri); \
>                                       Billing-Party=$avp($billing_party);
> \
>                                       Divert-Reason=$avp(s:divert_reason);
> \
>                                       X-RTP-Stat=$avp(s:rtp_statistics); \
>                                       From-Header=$hdr(from); \
>                                       User-Agent=$hdr(user-agent); \
>                                       Contact=$hdr(contact); \
>                                       Event=$hdr(event)")
> #
> SIP-Proxy-IP=$avp(s:sip_proxy_ip)")
>
>
>
> # -- group_radius params --
> modparam("group_radius", "use_domain", 1)
>
> # -- avpops params --
> #modparam("avpops", "avp_aliases", "day=i:101;time=i:102")
> modparam("avpops","avp_aliases","can_uri=i:34")
> modparam("avpops","avp_aliases","billing_party=i:1")
>
>
> # -- rr params --
> # add value to ;lr param to make some broken UAs happy
> modparam("rr", "enable_full_lr", 1)
>
> modparam("mediaproxy", "natping_interval", 60)
> modparam("registrar",  "nat_flag",         2)
>
> # -------------------------  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;
>        };
>
>
> # setflag(ACCOUNTING_FLAG);
> #       avp_write("SER_IP","$avp(s:sip-proxy)");
> #       avp_write("$ru", "$avp(can_uri)");
>
>        # 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");
>                if(is_method("BYE"))
>                { # log it all the time
>                    acc_rad_request("200 ok");
>                    acc_log_request("200 ok");
>                    setflag(1);
>                }
>
>                route(1);
>        };
>       if (src_ip==193.36.177.227) {
>                   fix_nated_sdp("2");
>       };
>        if(is_method("INVITE") && !has_totag())
>        {   # set the acc flags
>                setflag(1);
>                setflag(2);
>        };
>        if (method=="MESSAGE") {
>                 setflag(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") {
>                       # Uncomment this if you want to use digest
> authentication
> #                       if (!www_authorize("sip.com", "subscriber")) {
> #                               www_challenge("sip.com", "0");
> #                               exit;
> #                       };
>
> if (!radius_www_authorize(""))
>   {
>     www_challenge("","1");
>        exit;
> }
>            if (client_nat_test("3")) {
>                setflag(2);
>                force_rport();
>                fix_contact();
>            };
>
>                        save("location");
>                        exit;
>
>                };
>
>                lookup("aliases");
>                if (!uri==myself) {
>                        append_hf("P-hint: outbound alias\r\n");
>                        route(1);
>                };
>
>                # native SIP destinations are handled using our USRLOC DB
>                if (!lookup("location")) {
>            # log to acc as missed call
>                  acc_rad_request("404 Not Found");
>                  acc_log_request("404 Not Found");
>
>                  sl_send_reply("404", "Not Found");
>                  exit;
>                };
>        };
>
>    if (method=="INVITE") {
>        t_on_failure("1");
>    } else if (method == "BYE" || method == "CANCEL") {
>        end_media_session();
>    };
>
>    if (loose_route()) {
>        if (method=="INVITE" || method=="ACK") {
>            use_media_proxy();
>        };
> #if ((method=="INVITE" || method=="ACK") &&
> !to_uri=="sip:frog1 at xxx.xxx.10.12") {
> #            use_media_proxy();
> #};
>        t_relay();
>        return;
>    };
>
>    if (client_nat_test("3") && !search("^Record-Route:")) {
>        # Mark as NAT'ed
>        force_rport();
>        fix_contact();
>    };
>
>    if (method=="INVITE") {
>        t_on_reply("1");
>    };
>
>    if (method=="INVITE" || method=="ACK") {
>        use_media_proxy();
>    };
> #if ((method=="INVITE" || method=="ACK") &&
> !to_uri=="sip:frog1 at xxx.xxx.10.12") {
> #            use_media_proxy();
> #};
>
>    if (!t_relay()) {
>        if (method=="INVITE" || method=="ACK") {
>            end_media_session();
>        };
>        sl_reply_error();
>    };
>
>        append_hf("P-hint: usrloc applied\r\n");
>        # route(1);
>
> }
>
> route[1]
> {
>        # send it out now; use stateful forwarding as it works reliably
>        # even for UDP2TCP
>        if (!t_relay()) {
>                sl_reply_error();
>        };
>        exit;
> }
>
> failure_route[1] {
>    end_media_session();
> }
>
> onreply_route[1] {
>    if (status=~"(183)|(2[0-9][0-9])") {
>        if (client_nat_test("1")) {
>            fix_contact();
>        };
>        use_media_proxy();
>    };
> }
>
>
> --
> View this message in context:
> http://www.nabble.com/Openser-%2B-Mediaproxy-not-working-with-2-domains-tp14927928p14927928.html
> Sent from the OpenSER Users Mailing List mailing list archive at
> Nabble.com.
>
>
> _______________________________________________
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