[OpenSIPS-Users] OpenSIPS with public/private interface and RTPProxy
Răzvan Crainea
razvan at opensips.org
Thu May 9 14:40:35 CEST 2013
As also detailed in the other ticket, as well as in the documentation,
the engage_rtp_proxy() function has an undefined behavior when using in
a bridged scenario. Therefore I recommend you to use the
rtpproxy_offer/answer() functions.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/09/2013 02:34 PM, qasimakhan at gmail.com wrote:
> For engage_rtpproxy there are two flags that are used i.e. i for LAN
> interface and E for WAN interface. you can use these two flags to
> specify your direction of bridging. e.g. ie for LAN to WAN bridging and
> ei for WAN to LAN bridging. Meanwhile look at this documentation for
> detailed flag usage.
>
> http://www.opensips.org/html/docs/modules/1.8.x/rtpproxy.html#id292744
>
>
> Regards,
> Qasim
>
>
> On Thu, May 9, 2013 at 4:09 PM, Michele Pinassi
> <michele.pinassi at unisi.it <mailto:michele.pinassi at unisi.it>> wrote:
>
> Hi all,
>
> i have an OpenSIPS server with two interface, PUBLIC (xxx) and PRIVATE
> (172.20.1.2). The PRIVATE interface works inside a LAN dedicated to
> VoIP, with a MediaServer (172.20.1.5) and a Patton Gateway for PSTN
> (172.20.1.4).
>
> Users phone's can register on both interface and i use RTPProxy (in
> bridging mode) to ensure that both side can talk togheter.
>
> But something don't work as expected....
>
> Here's my OpenSIPS routing logic:
>
> ===========================================
> route{
> 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;
> };
>
> if(is_method("INVITE") && has_totag()) {
> engage_rtp_proxy();
> }
>
> if (has_totag()) {
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> if (is_method("BYE")) {
> setflag(1); # do accounting ...
> setflag(3); # ... even if the
> transaction fails
> } else if (is_method("INVITE")) {
> # even if in most of the cases is
> useless, do RR for
> # re-INVITEs alos, as some buggy
> clients do change route set
> # during the dialog.
> record_route();
> }
> # route it out to whatever destination was
> set by loose_route()
> # in $du (destination URI).
> route(1);
> } else {
> /* uncomment the following lines if you
> want to enable presence */
> if (is_method("SUBSCRIBE") && $rd ==
> "voip.unisi.it <http://voip.unisi.it>") {
> # in-dialog subscribe requests
> route(2);
> exit;
> }
> if ( is_method("ACK") ) {
> if ( t_check_trans() ) {
> # non loose-route, but
> stateful ACK; must be an ACK after
> # a 487 or e.g. 404 from
> upstream server
> t_relay();
> exit;
> } else {
> # ACK without matching
> transaction ->
> # ignore and discard
> exit;
> }
> }
> sl_send_reply("404","Not here");
> }
> exit;
> }
>
> #initial requests
>
> # CANCEL processing
> if (is_method("CANCEL"))
> {
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> t_check_trans();
>
> # authenticate if from local subscriber (uncomment to
> enable auth)
> # authenticate all initial non-REGISTER request that
> pretend to be
> # generated by local subscriber (domain from FROM URI is local)
> # if (!(method=="REGISTER") && from_uri==myself) /*no
> multidomain version*/
> if (!(method=="REGISTER") && is_from_local())
> /*multidomain version*/
> {
> if(!check_source_address("0")){
> if (!proxy_authorize("", "subscriber")) {
> proxy_challenge("", "0");
> exit;
> }
> if (!db_check_from()) {
> sl_send_reply("403","Forbidden auth ID");
> exit;
> }
>
> consume_credentials();
> # caller authenticated
> }
> }
>
> # preloaded route checking
> if (loose_route()) {
> xlog("L_ERR", "Attempt to route with preloaded Route's
> [$fu/$tu/$ru/$ci]");
> if (!is_method("ACK"))
> sl_send_reply("403","Preload Route denied");
> exit;
> }
>
> # record routing
> if (!is_method("REGISTER|MESSAGE"))
> record_route();
>
> # account only INVITEs
> if (is_method("INVITE")) {
> setflag(1); # do accounting
> }
> if (!uri==myself) {
> append_hf("P-hint: outbound\r\n");
> route(1);
> }
>
> if( is_method("PUBLISH|SUBSCRIBE")) {
> route(2);
> }
>
> if (is_method("REGISTER"))
> {
> # authenticate the REGISTER requests (uncomment to
> enable auth)
> if (!www_authorize("", "subscriber"))
> {
> www_challenge("", "0");
> exit;
> }
>
> if (!db_check_to())
> {
> sl_send_reply("403","Forbidden auth ID");
> exit;
> }
>
> if (!save("location"))
> sl_reply_error();
>
> exit;
> }
>
> if ($rU==NULL) {
> # request with no Username in RURI
> sl_send_reply("484","Address Incomplete");
> exit;
> }
>
> # media service number? (digits starting with *)
> if($rU=~"^\*") {
> route(4);
> }
>
> # apply DB based aliases (uncomment to enable)
> alias_db_lookup("dbaliases");
>
> # do lookup with method filtering
> if (!lookup("location","m")) {
> switch ($retcode) {
> case -1: # no contact: route it !
> cr_user_carrier("$fU", "$fd",
> "$avp(carrier)");
> if($avp(carrier)==0) {
> xlog("L_INFO","Not here:
> default route [$fd/$fu/$rd/$ru/$si]\n");
> # Not here: default route
> [172.20.1.4/sip:2425 at 172.20.1.4:5060/voip.unisi.it/sip:50 at voip.unisi.it:5060/172.20.1.4
> <http://172.20.1.4/sip:2425@172.20.1.4:5060/voip.unisi.it/sip:50@voip.unisi.it:5060/172.20.1.4>]
> if($(rU{s.len}) < 4) {
> xlog("L_ERR", "Number
> incomplete/failure for $rU\n");
> prefix("FAIL_");
> route(4);
> }
>
> if(!cr_route("default", "$fd",
> "$rU", "$rU", "call_id",
> "$avp(host)")) {
> xlog("L_ERR", "Number not
> found for $rU\n");
> prefix("FAIL_");
> route(4);
> }
> } else {
> xlog("L_INFO","Not here: user route
> [$fd/$fu/$rd/$ru/$si/$avp(carrier)]\n");
> $avp(domain)="voip.unisi.it
> <http://voip.unisi.it>";
> if (!cr_route("$avp(carrier)",
> "$avp(domain)", "$rU",
> "$rU","call_id", "$avp(host)")) {
> sl_send_reply("404", "Not
> found");
> xlog("L_ERR", "cr_route
> failed\n");
> exit;
> }
> }
> t_on_failure("1");
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> case -3: # internal error
> t_newtran();
> t_reply("404", "Not Found");
> exit;
> case -2: # method not supported
> sl_send_reply("405", "Method Not
> Allowed");
> exit;
> }
> }
>
> # when routing via usrloc, log the missed calls also
> setflag(2);
>
> route(1);
> }
>
> route[1] {
> xlog("L_INFO","Route1 [$fd/$fu/$rd/$ru/$si/]\n");
>
> # for INVITEs enable some additional helper routes
> if (is_method("INVITE")) {
> t_on_branch("2");
> t_on_reply("2");
> t_on_failure("1");
> }
>
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
>
> # Presence route
> route[2] {
> xlog("L_INFO","Route2 [$fd/$fu/$rd/$ru/$si/]\n");
>
> if (!t_newtran()) {
> sl_reply_error();
> exit;
> };
>
> if(is_method("PUBLISH")) {
> handle_publish();
> } else if( is_method("SUBSCRIBE")) {
> handle_subscribe();
> }
>
> exit;
> }
>
> route[4] {
> xlog("L_INFO","Route4 [$fd/$fu/$rd/$ru/$si/]\n");
>
> rewritehostport("172.20.1.5:5060 <http://172.20.1.5:5060>");
> route(1);
> }
>
> branch_route[2] {
> xlog("L_INFO","Branch Route2 [$fd/$fu/$rd/$ru/$si/]\n");
> }
>
> onreply_route[2] {
> xlog("L_INFO","OnReply Route2 [$fd/$fu/$rd/$ru/$si/]\n");
> }
>
> failure_route[1] {
> xlog("L_INFO","Failure Route1 [$fd/$fu/$rd/$ru/$si/]\n");
>
> if (t_was_cancelled()) {
> exit;
> }
>
> if (t_check_status("408|5[0-9][0-9]")) {
> if(!cr_route("default", "$fd", "$rU", "$rU", "call_id",
> "$avp(host)")){
> t_reply("403", "Not allowed");
> } else {
> t_on_failure("2");
> t_relay();
> }
> }
> }
>
> failure_route[2] {
> xlog("L_INFO","Failure Route2 [$fd/$fu/$rd/$ru/$si/]\n");
>
> if (t_was_cancelled()) {
> exit;
> }
>
> revert_uri();
> prefix("FAILURE_");
> rewritehostport("172.20.1.5:5060 <http://172.20.1.5:5060>");
> t_relay();
> }
> ===========================================
>
> and this is the RTPProxy config:
>
> ===========================================
> CONTROL_SOCK=udp:127.0.0.1:12221 <http://127.0.0.1:12221>
>
> # Additional options that are passed to the daemon.
> EXTRA_OPTS="-l [external IP]/172.20.1.2 <http://172.20.1.2>"
> ===========================================
>
> Anybody can help me ?
>
> Michele
>
> --
> Michele Pinassi
> Responsabile Telefonia di Ateneo
> Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi
> di Siena
> tel: 0577.(23)2169 <tel:0577.%2823%292169> - fax: 0577.(23)2053
> <tel:0577.%2823%292053>
>
> Per trovare una soluzione rapida ai tuoi problemi tecnici
> consulta le FAQ di Ateneo, http://www.faq.unisi.it
>
>
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>
>
>
>
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