[OpenSIPS-Users] NAT problem, no-audio when calling outside network... Please help
Khan
khansfriend at gmail.com
Sun Aug 30 02:27:12 CEST 2009
Hey everyone,
I have been trying to work this for a long time, this mailing list is
my last resort. I have applied NAT traversal using RTP proxy. My
scenario is as follows:
UAC1 (behind NAT) ---> UAC2 (behind NAT)
The UAC's get authenticated fine, call establishes but there is no
voice, neither i hear them nor they hear me. I can't pin point exactly
where did i go wrong. My script is as follows:
route{
## unrelated script has been stripped!!!
if (nat_uac_test("3")) {
if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
log("LOG:Someone trying to register from private IP, rewriting\n");
# Rewrite contact with source IP of signalling
fix_nated_contact();
if ( is_method("INVITE") ) {
fix_nated_sdp("1"); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setbflag(6); # Mark as NATed
# if you want sip nat pinging
setbflag(8);
xlog("L_INFO", "fixNATed and setbflag 6, 8 - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
};
};
# sequential requests...
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
xlog("L_INFO", "Initial loose-routing - M=$rm RURI=$ru F=$fu T=$tu
IP=$si \n");
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
xlog("L_INFO", "BYE ... unforce RTP - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
unforce_rtp_proxy();
} 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 $du (destination URI).
route(1);
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
#initial requests
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
xlog("L_INFO", "CANCEL ... unforce RTP - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
unforce_rtp_proxy();
exit;
}
#--> Preventing the UAC problem which sends Option
##if(is_method("OPTIONS")) {
## sl_send_reply("200", "OK");
## exit;
##}
#--> uncommented followings
if ((method=="OPTIONS|SUBSCRIBE") && from_uri==myself) /*no
multidomain version*/
##if (!(method=="OPTIONS") && is_from_local()) /*multidomain version*/
{
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
# caller authenticated
}
t_check_trans();
if (!(method=="REGISTER") && from_uri==myself) /*no multidomain version*/
##if (!(method=="REGISTER") && is_from_local()) /*multidomain version*/
{
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
# caller authenticated
}
# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
if (!uri==myself)
## replace with following line if multi-domain support is used
##if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
##if($rd=="tls_domain1.net") {
## t_relay("tls:domain1.net");
## exit;
##} else if($rd=="tls_domain2.net") {
## t_relay("tls:domain2.net");
## exit;
##}
route(1);
}
# requests for my domain
if (is_method("PUBLISH")) {
sl_send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER")) {
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("", "subscriber")) {
xlog("L_INFO", "1st Pass - Register authentication - M=$rm RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
www_challenge("", "0");
exit;
}
if (!check_to()) {
xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
xlog("L_INFO", "2nd Pass - Registration successful - M=$rm RURI=$ru
F=$fu T=$tu IP=$si ID=$ci\n");
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
if (!lookup("location")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
#------>
route[1] {
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
exit;
};
if (isbflagset(6)) {
force_rtp_proxy();
};
t_on_reply("1");
#! *** <<
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
*************************************************************************
The output capture from WireShark is at the following link.
http://pastebin.com/m1c17484d
Please help me figure out this problem, I appreciate your time.
Thank you,
Khan
VoIP Rookie
Every beginning has an end regardless we believe it or not...
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