17 Jan
2008
17 Jan
'08
8:43 p.m.
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:openserrw@localhost/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:frog1@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 ((method=="INVITE" || method=="ACK") &&
!to_uri=="sip:frog1@xxx.xxx.10.12") {
# 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();
};
}
--
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18 Jan
18 Jan
12:36 p.m.
New subject: [OpenSER-Users] Openser + Mediaproxy not working with 2 domains
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 <mbedial(a)gmail.com>om>:
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:openserrw@localhost/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:frog1@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 ((method=="INVITE" || method=="ACK") &&
!to_uri=="sip:frog1@xxx.xxx.10.12") {
# 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-tp1…
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Nabble.com.
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
21 Jan
21 Jan
7:52 p.m.
New subject: [OpenSER-Users] Openser + Mediaproxy not working with 2 domains
2008/1/17, Andreti <mbedial(a)gmail.com>om>:
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@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();
};
}
5964
days inactive
5968
days old
2 comments
2 participants
participants (2)
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Andreti -
Mario Bedialauneta