[OpenSIPS-Users] RTPProxy to link two networks
Michele Pinassi
michele.pinassi at unisi.it
Fri Oct 18 15:02:33 CEST 2013
Hi all,
i have Opensips on a server with two interfaces, PUBLIC and PRIVATE
(172.20.x.x).
This is my opensips.cfg:
####### Routing Logic ########
route {
force_rport();
if (is_method("INVITE")) {
if (!get_source_group("$var(group)") ) {
$var(group) = 3; /* Default value */
}
# User group:
# 1 = Internal VoIP subnet 172.20
# 2 = External VoIP subnet but inside Unisi context 172.16 | 10.0.0
# 3 = Outside/foreign (not allowed)
xlog("L_INFO", "User group is $var(group) [$fd/$fu/$rd/$ru/$si]\n");
if ($var(group) == 2) { # Outside VoIP LAN
if (is_method("REGISTER")) {
fix_nated_register();
setbflag(NAT);
} else {
fix_nated_contact();
setflag(NAT);
}
}
}
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
# Verifica che l'IP provenga da una rete abilitata (address table)
if (!check_address("0","$si","$sp","$proto")) {
xlog("L_INFO","IP $si Forbidden");
sl_send_reply("403", "Forbidden");
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
# validate the sequential request against dialog
if ( $DLG_status!=NULL && !validate_dialog() ) {
xlog("In-Dialog $rm from $si (callid=$ci) is not valid
according to dialog\n");
## exit;
}
if (is_method("BYE")) {
setflag(ACC_DO); # do accounting ...
setflag(ACC_FAILED); # ... 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();
}
if (check_route_param("nat=yes"))
setflag(NAT);
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(relay);
} else {
if (is_method("SUBSCRIBE") && $rd == "127.0.0.1:5060") { #
CUSTOMIZE ME
# in-dialog subscribe requests
route(handle_presence);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK
after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
if ( !(is_method("REGISTER") || is_from_gw() ) ) {
if (is_from_local()) {
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!db_check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
} else {
if (!is_uri_host_local()) {
send_reply("403","Rely forbidden");
exit;
}
}
}
# preloaded route checking
if (loose_route()) {
xlog("L_ERR", "Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
# create dialog with timeout
if ( !create_dialog("B") ) {
send_reply("500","Internal Server Error");
exit;
}
setflag(ACC_DO); # do accounting
}
if (!is_uri_host_local()) {
append_hf("P-hint: outbound\r\n");
route(relay);
}
# requests for my domain
if( is_method("PUBLISH|SUBSCRIBE"))
route(handle_presence);
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
if (!db_check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if ( 0 ) setflag(TCP_PERSISTENT);
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases
alias_db_lookup("dbaliases");
# do lookup with method filtering
if (!lookup("location","m")) {
switch ($retcode) {
case -1: # Note here: check for blacklist numbers
if (!check_blacklist("globalblacklist")) {
sl_send_reply("403", "Forbidden");
exit;
}
# Check for user ACLs
if($rU=~"^\0") {
if(!db_is_user_in("Credentials","extern")) {
xlog("L_INFO","403 - Forbidden
[$fd/$fu/$rd/$ru/$si]\n");
sl_send_reply("403", "Forbidden - No permission for
external calls");
exit;
}
}
# Then route it !
cr_user_carrier("$fU", "$fd", "$avp(carrier)");
if($avp(carrier)==0) {
xlog("L_INFO","Not here: default route
[$fd/$fu/$rd/$ru/$si]\n");
if($(rU{s.len}) < 4) {
xlog("L_ERR", "Number incomplete/failure for $rU\n");
prefix("FAIL_");
route(vmbox);
}
if(!cr_route("default", "$fd", "$rU", "$rU",
"call_id", "$avp(host)")) {
xlog("L_ERR", "Number not found for $rU\n");
prefix("FAIL_");
route(vmbox);
}
} else {
xlog("L_INFO","Not here: user route
[$fd/$fu/$rd/$ru/$si/$avp(carrier)]\n");
$avp(domain)="voip.unisi.it";
if (!cr_route("$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("missed_call");
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;
}
}
if (isbflagset(NAT)) setflag(NAT);
# when routing via usrloc, log the missed calls also
setflag(ACC_MISSED);
route(relay);
}
route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
if (isflagset(NAT)) {
rtpproxy_offer("ro");
}
t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}
if (isflagset(NAT)) {
add_rr_param(";nat=yes");
}
if (!t_relay()) {
send_reply("500","Internal Error");
};
exit;
}
# Presence route
route[handle_presence]
{
if (!t_newtran())
{
sl_reply_error();
exit;
}
if(is_method("PUBLISH"))
{
handle_publish();
}
else
if( is_method("SUBSCRIBE"))
{
handle_subscribe();
}
exit;
}
branch_route[per_branch_ops] {
xlog("new branch at $ru\n");
}
onreply_route[handle_nat] {
if (nat_uac_test("1"))
fix_nated_contact();
if ( isflagset(NAT) )
rtpproxy_answer("ro");
# xlog("incoming reply\n");
}
failure_route[missed_call] {
if (t_was_cancelled()) {
exit;
}
if (t_check_status("408|5[0-9][0-9]")) {
if(!cr_route("default", "$fd", "$rU", "$rU", "call_id",
"$avp(host)")){
t_reply("403", "Not allowed");
} else {
revert_uri();
prefix("FAILURE_");
rewritehostport("172.20.1.5:5060");
t_relay();
}
}
}
route[vmbox] {
xlog("L_INFO","Route VMBOX [$fd/$fu/$rd/$ru/$si/]\n");
rewritehostport("172.20.1.5:5060");
route(relay);
}
local_route {
if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
acc_db_request("200 Dialog Timeout", "acc");
}
}
SIP Signaling works but RTP (Voice) not. If i register a device like
"sip:5001 at 10.1.65.117" call signalling works but voice not.
Any hint ?
Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)2169 - fax: 0577.(23)2053
Per trovare una soluzione rapida ai tuoi problemi tecnici
consulta le FAQ di Ateneo, http://www.faq.unisi.it
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