RE: [Serusers] fr_inv_timer problem when using SIPPS - sr-users

18 Nov 2003

Is the c-timer of RFC3261 the same as the fr_inv_timer? Then you're
right:

RC3261
" Update timer C for provisional responses

         For an INVITE transaction, if the response is a provisional
         response with status codes 101 to 199 inclusive (i.e., anything
         but 100), the proxy MUST reset timer C for that client
         transaction.  The timer MAY be reset to a different value, but
         this value MUST be greater than 3 minutes
"

But setting the timer to 3 minutes is much to long for redirection to
voicemail. 

Shouldn't timer-c be activated for all transactions (as stated in the
RFC) and not only for calls which will be redirected to the voicemail?
Therefore, a separate timer for redirection to voicemail would be
necessary. The fr_inv_timer would be set to 3 minutes to cancel
neverending transactions, and a separate fr_inv_voicemail_timer can be
set to e.g. 10 seconds and this timer will not be reset after
provisional responses.

regards, 
Klaus

...
  -----Original Message-----
 From: Jiri Kuthan [mailto:jiri@iptel.org] 
 Sent: Tuesday, November 18, 2003 12:47 PM
 To: Klaus Darilion; serusers(a)lists.iptel.org
 Subject: Re: [Serusers] fr_inv_timer problem when using SIPPS

 Thanks, it is very nice to get some real questions :)

 I'm now running in my bed, so I may be confused but I think that
 the spec tells 180s to extend life of transaction context (i.e.,
 extend C-timer). I don't necessarily like it, and I think it was 
 just a too quick result of a discussion which the sip mailing list 
 wanted to complete (roughly during bis-07 time).

 I need to review what SER actually does. Not extending the transaction
 lifetime through 18x should be at least a config option.

 -jiri

 At 12:00 PM 11/18/2003, Klaus Darilion wrote:
 Hi!

I tried voicemail with redirect to the vm-proxy if the user   doesn't pick up
 the phone after 10 seconds. It works fine if I
call a kphone   user, but not
 with SIPPS users. SIPPS sends a 180 Ringing every
five seconds which
probably resets the timer - if I choose timer intervalls   shorter than 5
 seconds it also works with SIPPS.

Is this a bug?

tried with ser:
0.8.12dev-t14
0.8.11rc1
0.8.11-r1

regards,
Klaus

----------
#
# $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=no # (cmd line: -E)

/* 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"
alias=obelix.ict.tuwien.ac.at
# ------------------ 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"

# 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   ---------------

# -- 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",
"sql://ser:ser2000@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",
"sql://ser:ser2000@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","sql://ser:ser2000@localhost/ser")

# -- voicemail params --
modparam("group",
"db_url","sql://serro:serro2000@localhost/ser")

# -- nathelper params --
#modparam("nathelper", "natping_interval", 10)

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

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

# main routing logic

route{

        lookup("aliases");
        # 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;
        };

        # 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

#remove record route for dissipate
#if (!(method=="REGISTER")) record_route();

        # loose-route processing
        if (loose_route()) {
                t_relay();
                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("obelix.ict.tuwien.ac.at",
"subscriber")) {
#                                
www_challenge("obelix.ict.tuwien.ac.at",
 "0");
#                               break;
#                       };

                        if (!save("location")) {
                                sl_reply_error();
                        };
                        break;
                };

                # check if number beginns with 00, then   convert it into a +
                 if
(uri=~"^sip:00[0-9]*@") {
                        # strip booth leading "0"
                        strip(2);
                        prefix("+");
                };
                # check if request uri begins with an   internation phone
 number, if yes, try enum to resolve
                if (uri=~"sip:\+?[0-9]+@.*") {
                        enum_query("voice");
                };

                #mark transaction for voicemail
                if (is_user_in("Request-URI", "voicemail")) {
                        setflag(4);
                };

                # native SIP destinations are handled using   our USRLOC DB
                 if (!lookup("location"))
{
                        # handle user which was not found
                        route(4);
                        break;
                };
        };

        #add failure route which should be performed if   response code >=300
         if  (method=="INVITE" &&
isflagset(4)) {
                t_on_failure("1");
        };

        # forward to current uri now; use stateful forwarding; that
        # works reliably even if we forward from TCP to UDP
        if (!t_relay()) {
                sl_reply_error();
        };

}

route[4]{

        # non-Voip -- just send "off-line"
        if (!(method == "INVITE" || method == "ACK" ||   method
== "CANCEL" ||
 method == "REFER" || method ==
"BYE")) {
                sl_send_reply("404", "Not Found");
                break;
        };

        # not voicemail subscriber
        if (!isflagset(4)) { 
                sl_send_reply("404", "Not Found and no   voicemail
turned
 on");
                break;
        };

        # forward to voicemail now
        #rewritehostport("machtnix.ict.tuwien.ac.at:5060");
        t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");

        #klaus
        #forward(machtnix.ict.tuwien.ac.at, 5060)
}

failure_route[1] {
  /* XX: note: unsafe if preloaded routes without username used */
  revert_uri();
        revert_uri();
        #rewritehostport("machtnix.ict.tuwien.ac.at:5060");
        append_branch();
        t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");
}

_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers  
 --
 Jiri Kuthan            http://iptel.org/~jiri/ 