Hi
anyone can help mi with SIP (Linux) UDP -> TCP conversion and vice versa
using SER? i am having too large UDP SIP packets fragmented. it there
any other free source out there that can do such conversion?
Cheers
Goh
shengmai_goh(a)innomedia.com.sg
Goh Sheng Mai wrote:
hi
i have large voip udp packets over internet problems and i need the
transfer medium to be tcp. below is what i had in mind.
SIPp_client A (TCP out) -> (listen TCP) SER 1 (UDP out) --> Soft-switch
(UDP in /out only) --> (to) ring Client B (listen UDP)
Client A and Client B registers to above Soft-switch. All the above are
individual servers (Linux CentOS based)
i need help in a (conversion) ser.cfg. Can anyone post a sample ?? below
is my ser.cfg (tried using t_relay_to_udp(), rewritehost(),
rewritehostport(), forward functions BUT none seems to work the way i
want it to)
SER 1 == 192.168.10.203
Softswitch == 192.168.10.251
Client A == 192.168.10.221 or .236 (SIPp client-registered to
soft-switch successfully)
Client B == 192.168.x.x (registered to soft-switch successfully)
Thanks
shengmai
shengmai_goh(a)innomedia.com.sg
#
# $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#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/print.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/local/lib/ser/modules/auth.so"
#loadmodule "/usr/local/lib/ser/modules/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)
# -- 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)
# ------------------------- 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");
break;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
break;
};
# 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();
#if (method == "INVITE") {
# if (uri =~ "sip:*@*"){
# log(1, "Forwarding to Vsp12K\n");
# rewritehostport("192.168.10.204:5060");
# t_relay();
# break;
# }
#};
if (uri =~ "sip:*@*"){
#log(1, "Forwarding to Vsp12K\n");
#rewritehostport("192.168.10.204:5060");
if (uri =~ "sip:*@192.168.10.203"){
rewritehostport("192.168.10.251:5060");
}
t_relay_to_udp("192.168.10.251", "5060");
#t_relay();
break;
};
# 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("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# -----------------------------------------------------------------
# Pass on stuff going to PSTN to Asterisk
# -----------------------------------------------------------------
#if (uri=~"^sip:*@.*") {
# rewritehostport ("*192.168.10.204:5060");
# if (!t_relay()) {
# # sl_send_reply ("403", "prout");
# sl_reply_error();
# };
# break;
#};
rewritehostport("192.168.10.204:5060");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
break;
}
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