[Serusers] SER+Rtp proxy -> no voice going through

[Tareq Siraj]

Hi all, I have found the following ser.cfg file for
SER+RTP proxy combination in the mailing list.
However, with this, no audio is going through. The
call goes through, if i accept the call, no audio. I'm
completely new to SER and SIP servers and still going
through the manuals for the ser script. Any help will
be appreciated. Thanks.

- tareq

#
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#

# ----------- global configuration parameters
------------------------

debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)

listen=192.168.1.21

# hostname matching an alias will satisfy the
condition uri==myself".
alias=mypc
alias=mypc.localdomain


# Uncomment these lines to enter debugging mode
#debug=7
#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
----------------------------------

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"

# 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"

# load the voicemail module
#loadmodule "/usr/local/lib/ser/modules/vm.so"

# load the enum module
loadmodule "/usr/local/lib/ser/modules/enum.so"

# load the group module, to verify if a user forwards
to voicemail
loadmodule "/usr/local/lib/ser/modules/group.so"

# load the nathelper module
loadmodule "/usr/local/lib/ser/modules/nathelper.so"

# ----------------- setting module-specific parameters
---------------

# -- registrar parameter
# special NAT flag indicates that a registered client
is behind NAT
modparam("registrar", "nat_flag", 6)

# -- 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", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:heslo@localhost/ser")

# -- 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")
modparam("auth_db", "db_url",
"mysql://ser:heslo@localhost/ser")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

# -- voicemail params --
#modparam("voicemail",
"db_url","mysql://login:password@localhost/ser")

# -- voicemail params --
#modparam("group",
"db_url","mysql://login:password@localhost/ser")

# -- nathelper params --
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)

modparam("tm", "fr_inv_timer", 30 )
#modparam("tm", "fr_inv_timer", 8 )

#Explicitly set the socket used by rtpproxy
#modparam("nathelpler", "rtpproxy_sock",
"/var/run/rtpproxy.sock")


# ------------------------- request routing logic
-------------------

# main routing logic

route{
log(1,
"-------------------------------------------\n");
log(1, "entering main loop\n");

if (nat_uac_test("2")) {
log(1, "src address different than via header->NAT
detected\n");
log(1, "force_rport and fix_nated_contact and
setflag(5)\n");
#try NAT traversal, works only if the client is
symmetrical
force_rport();
fix_nated_contact();
append_hf("P-hint: fixed NAT contact for
request\r\n");
# flag 5 indicates that incoming request is from NATed
client
setflag(5);
};

if (method=="REGISTER")
log(1, "REGISTER message received\n");

if (method=="INVITE")
log(1, "INVITE message received\n");

if (method=="ACK")
log(1, "ACK message received\n");

if (method=="BYE")
log(1, "BYE message received\n");

if (method=="CANCEL")
log(1, "CANCEL message received\n");

if (method=="SUBSCRIBE")
log(1, "SUBSCRIBE message received\n");

if (method=="NOTIFY")
log(1, "NOTIFY message received\n");

if (method=="OPTIONS")
log(1, "OPTIONS message received\n");

if (method=="INFO")
log(1, "INFO message received\n");

if (method=="MESSAGE")
log(1, "MESSAGE message received\n");

if (method=="REFER")
log(1, "REFER message received\n");

# 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) {
#if (len_gt( max_len )) {
sl_send_reply("513", "Message too big");
break;
};

# loose-route processing
if (loose_route()) {
log(1, "loose_route processing\n");
t_relay();
break;
};

# create transaction state; abort if error occured
# if ( !t_newtran()) {
# sl_reply_error();
# break;
# };

#new
# now check if it's about PSTN destinations through
our gateway;
# note that 8.... is exempted for numerical non-gw
destinations
if (uri=~"^sip:0[0-9]*@.*") {
log(1, "PSTN call...\n");
route(3);
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") {
log(1, "analyzing REGISTER request\n");
# Uncomment this if you want to use digest
authentication
if (!www_authorize("mypc", "subscriber")) {
www_challenge("mypc", "0");
break;
};

if (isflagset(5)) {
#register from nated client, save nat_flag=6
#in location table
setflag(6);
};
if (!save("location")) {
log(1, "save location error\n");
sl_reply_error();
};
break;
};

lookup("aliases");


#mark transaction for voicemail
if (is_user_in("Request-URI", "voicemail\n")) {
log(1, "requested user is in voicemail group");
setflag(4);
};
# native SIP destinations are handled using our USRLOC
DB
if (!lookup("location")) {
# handle user which was not found
log(1, "requested user not found\n");
route(4);
break;
};
};

#add failure route which should be performed if
response code >=300
if (method=="INVITE" && isflagset(4)) {
log(1, "invite for voicemail user->initiate
failureroute[1]\n");
t_on_failure("1");
};

# forward to current uri now; use stateful forwarding;
that
# works reliably even if we forward from TCP to UDP

route(1);
}

route[1]{
log(1,
"-------------------------------------------\n");
log(1, "entering route[1] - relaying SIP message\n");
if ((isflagset(5)) || (isflagset(6))) {
log(1, "at least one of the participants is
NATed->record_route\n");
record_route();
log(1, " -->setting up reply processing
->onreply_route[1]");
t_on_reply("1");
if (method=="INVITE") {
log(1, " INVITE request-->force_rtp_proxy, set
NATED-INVITE flag(7)");
force_rtp_proxy();
append_hf("P-hint: request forced to rtp proxy\r\n");
setflag(7);
};
};

log(1, "relaying message ...\n");
if (!t_relay()) {
log(1, "t_relay error occured\n");
sl_reply_error();
};

}

# all incoming replies for t_onrepli-ed transactions
enter here
onreply_route[1] {
log(1,
"-------------------------------------------\n");
log(1, "onreply_route[1] entered\n");
if (isflagset(6)) {
log(1, "transaction was sent to a NATED client -> fix
nated contact\n");
fix_nated_contact();
append_hf("P-hint: fixed NAT contact for
response\r\n");
}

if ( (status=~"100") ) {
log(1, "status 100 received\n");
};

if ( (status=~"180") ) {
log(1, "status 180 received\n");
};

if ( (status=~"202") ) {
log(1, "status 202 received\n");
};

if ( (status=~"200" || status=~"183") ) {
log(1, "status 2xx or 183");
if ( isflagset(7) ) {
log(1, "marked(7) as NATED-INVITE -> force_rtp_proxy
\n");
force_rtp_proxy();
append_hf("P-hint: response forced to rtp proxy\r\n");
};
};
}

#new
# logic for calls to the PSTN
route[3] {
# turn accounting on
setflag(1);

/* require all who call PSTN to be members of the
"int" group;
apply ACLs only to INVITEs -- we don't need to protect
other requests, as they
don't imply charges; also it could cause troubles when
a call comes in via PSTN
and goes to a party that can't authenticate
(voicemail, other domain) -- BYEs would
fail then; exempt Cisco gateway from authentication by
IP address -- it does not
support digest
*/
if (method=="INVITE" && (!src_ip==192.168.1.21)) {
if (!proxy_authorize( "mypc" /* realm */,
"subscriber" /* table name */)) {
proxy_challenge( "mypc" /* realm */, "0" /* no qop */
);
break;
};
# let's check from=id ... avoids accounting confusion

if(!is_user_in("credentials", "int")) {
sl_send_reply("403", "NO PSTN Privileges...");
break;
};
consume_credentials();

}; # INVITE to authorized PSTN

# if you have passed through all the checks, let your
call go to GW!
force_rtp_proxy();
record_route();
t_on_reply("1");
# snom conditioner
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};

append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP
ephemeral)
#t_relay_to_udp("212.17.35.184","5060");
t_relay_to_udp("192.168.1.1","5060");
}



route[4]{
log(1,
"-------------------------------------------\n");
log(1, "entering route[4] = requested user not
online\n");
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method
== "CANCEL" || method == "REFER" || method == "BYE"))
{
log(1, "no invite,ack,cancel,refer->return 404\n");
sl_send_reply("404", "Not Found");
break;
};

# not voicemail subscriber and no echo/conference call
if ( isflagset(4)) {
log(1, "flag(4) active\n");
};
if (uri =~ "conference") {
log(1, "conference call\n");
};
if (uri =~ "echo") {
log(1, "echo call\n");
};
if ( !( isflagset(4) || (uri =~ "conference") || (uri
=~ "echo") ) ) {
log(1, "no voicemail subscriber->return 404");
sl_send_reply("404", "Not Found and no voicemail
turned on");
break;
};

if ( isflagset(5) ) {
log(1, "caller is NATed->record_route\n");
record_route();
log(1, " -->setting up reply processing
->onreply_route[1]");
t_on_reply("1");
if (method=="INVITE") {
log(1, " INVITE request-->force_rtp_proxy");
force_rtp_proxy();
};
};

# forward to voicemail now
² rewritehostport("192.168.1.21:5060");
log(1, "forward to voicemail\n");
t_relay_to_udp("192.168.1.21", "5060");

}



failure_route[1] {
/* XX: note: unsafe if preloaded routes without
username used */
log(1,
"-------------------------------------------\n");
log(1, "failureroute[1] entered\");
revert_uri();
rewritehostport("192.168.1.21:5060");
append_branch();
t_relay_to_udp("192.168.1.21", "5060");

}


		
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