Even if half of all RTP packets get lost
you will just have bad quality of call.
In your case it very looks like you have
problem with ACK
DO ngrep and see if PSTN GW receive ACK
packet after OK (otherwise GW will repeat OK few times and then disconnect
call) if it is an issue just do search in this list with keyword ACK
From: serusers-bounces@lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On Behalf
Of support
Sent: Saturday, February 26, 2005
12:24 AM
To: serusers@lists.iptel.org
Subject: [Serusers] ser + rtpproxy
- call disconnected
Hi,
When I try to make a call using rtpproxy
and ser-0.8.14 (SIP UA <---> PSTN), most of the time the call will be
disconnected within 1 minutes. Sometimes, it will be so unstable that the call be
disconnected after 15 seconds. The above scenarios happens
Will this problem related to unstable
rtpproxy? Because I know that all packets will route through the server. once
rtp packet get loss, the call will be disconnected. Is this correct?
Perhaps it depends on my ser.cfg.
Thomas
My ser.cfg:
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration
parameters ------------------------
debug=3
# debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd.
line: -r)
rev_dns=no # (cmd. line: -R)
listen=""
port=5060
children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
# ------------------ module loading
----------------------------------
loadmodule
"/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting
module-specific parameters ---------------
# -- usrloc params --
# Uncomment this if you want to use SQL
database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- 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")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- Nathelper params --
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) #
modparam("nathelper", "ping_nated_only", 1) #
# ------------------------- request
routing logic -------------------
# main routing logic
route{
#
-----------------------------------------------
# Sanity Check Section
# -----------------------------------------------
# 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#
-----------------------------------------------
# NOTIFY Keep-Alive Section
# -----------------------------------------------
if ((method=="NOTIFY") && search("^Event:
keep-alive")) {
sl_send_reply("200","OK");
break;
};
# Nathelper
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == "REGISTER" || !
search("^Record-Route:")) {
fix_nated_contact(); # Rewrite contact with
source IP of signalling
if (method == "INVITE") {
fix_nated_sdp("1");
# Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost
Via
setflag(6); # Mark as NATed
};
};
# 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");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
# 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("", "subscriber")) {
www_challenge("", "0");
break;
};
save("location");
break;
};
# if the dialed number lies
in the range 35891500-35891799, don't forward it to T1 Trunk GW
if ((uri=~"^sip:(852|)358915[0-9][0-9]@") ||
(uri=~"^sip:(852|)358916[0-9][0-9]@") ||
(uri=~"^sip:(852|)358917[0-9][0-9]@")) {
if (uri=~"^sip:852*") {
strip(3);
};
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC
# Call Routing Section
if (!lookup("location")) {
if (uri=~"^sip:(852|)[0-9]{8}@") {
# Send to PSTN Gateway
route(2);
break;
};
sl_send_reply("404", "User Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# Nathelper
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");
break;
};
# if client or server know to be
behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply
to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
break;
};
}
# PSTN Call to T1 Trunk GW
route[2] {
rewritehostport("");
if (isflagset(6)) {
force_rtp_proxy();
};
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
break;
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~
"(183)|2[0-9][0-9]") {
fix_nated_contact();
# Not all 2xx messages have a
content body so here we make sure
# out Content-Length > 0 to avoid a parse error
if (!search("^Content-Length:\0")) {
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();
};
}
---------end of config -----------