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 <[EMAIL PROTECTED]>:
>
>
> 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:[EMAIL PROTECTED]/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:[EMAIL PROTECTED]") {
> # 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:[EMAIL PROTECTED]") {
> # 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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>
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