[OpenSER-Users] Openser + Mediaproxy not working with 2 domains
Mario Bedialauneta
mbedial at gmail.com
Mon Jan 21 19:52:22 CET 2008
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
>
>
> # ------------------------- 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)");
>
> if (!method=="REGISTER") record_route();
>
> 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 (uri==myself) {
>
> if (method=="REGISTER") {
> # Uncomment this if you want to use digest
>
> 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 (!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();
> };
> }
>
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.kamailio.org/pipermail/users/attachments/20080121/3eb92293/attachment.htm
More information about the Users
mailing list