Kamailio as load balancer for multiple asterisk servers

Question

I have three virtual servers running Ubuntu 14.04. On one server I installed Kamailio and on the others Asterisk. I want that the Kamailio server works as a load balancer and forwards the incoming calls to the asterisk servers (round robin).

I want to test it first with one asterisk server and if that works, I can add more for more performance.

I added my SIP provider credentials like this:

kamctl add test testpasswd

Then I added the asterisk server to the dispatcher table like this:

INSERT INTO dispatcher (setid,destination,flags,priority,attrs,description) VALUES (1,"sip:10.1.1.3:5060",0,0,"","Asteriskl-I");

I changed the sip.conf file on my asterisk server that it connects to my kamailio server and that seems to work.

My kamailio.cfg file looks like this:

#!KAMAILIO
#
# sample config file for dispatcher module
# - load balancing of VoIP calls with round robin
# - no TPC listening
# - don't dispatch REGISTER and presence requests
#
# Kamailio (OpenSER) SIP Server v3.2
#     - web: http://www.kamailio.org
#     - git: http://sip-router.org
#
# Direct your questions about this file to: [email protected]
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
#     - define WITH_DEBUG
#

####### Global Parameters #########

#!define WITH_DEBUG

#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif

memdbg=5
memlog=5

log_facility=LOG_LOCAL0

fork=yes
children=4

/* comment the next line to enable TCP */
disable_tcp=yes

/* uncomment the next line to disable the auto discovery of local aliases
   based on revers DNS on IPs (default on) */
auto_aliases=no

/* add local domain aliases */
# alias="mysipserver.com"

port=5060

/* uncomment and configure the following line if you want Kamailio to
   bind on a specific interface/port/proto (default bind on all available) */
# listen=udp:127.0.0.1:5060

sip_warning=no

####### Modules Section ########

#set module path
mpath="/usr/local/lib64/kamailio/modules/"

# loadmodule "db_mysql.so"
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
# loadmodule "ctl.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
loadmodule "dispatcher.so"


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


# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")


# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)


# ----- acc params -----
modparam("acc", "log_flag", 1)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_extra",
        "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")

# ----- tm params -----
modparam("tm", "fr_timer", 2000)
modparam("tm", "fr_inv_timer", 40000)

# ----- dispatcher params -----
# modparam("dispatcher", "db_url",
#       "mysql://kamailio:123456@localhost/kamailio")
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")

####### Routing Logic ########


# main request routing logic

route {

        # per request initial checks
        route(REQINIT);

        # handle requests within SIP dialogs
        route(WITHINDLG);

        ### only initial requests (no To tag)

        # CANCEL processing
        if (is_method("CANCEL"))
        {
                if (t_check_trans())
                        t_relay();
                exit;
        }

        t_check_trans();

        # record routing for dialog forming requests (in case they are routed)
        # - remove preloaded route headers
        remove_hf("Route");
        if (is_method("INVITE|SUBSCRIBE"))
                record_route();
        #if (is_method("INVITE"))
        #{
        #       ds_select_domain("1","4");
        #       #sl_send_reply("300","Redirect");
        #       #t_relay();
        #       exit;
        #}

        # account only INVITEs
        if (is_method("INVITE"))
        {
                setflag(1); # do accounting
        }

        # handle presence related requests
        route(PRESENCE);

        # handle registrations
        route(REGISTRAR);

        if ($rU==$null)
        {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }

        # dispatch destinations
        route(DISPATCH);

        route(RELAY);
}


route[RELAY] {
        if (!t_relay()) {
                sl_reply_error();
        }
        exit;
}

# Per SIP request initial checks
route[REQINIT] {
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }

        if(!sanity_check("1511", "7"))
        {
                xlog("Malformed SIP message from $si:$sp\n");
                exit;
        }
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
        if (has_totag()) {
                # sequential request withing a dialog should
                # take the path determined by record-routing
                if (loose_route()) {
                        if (is_method("BYE")) {
                                setflag(1); # do accounting ...
                                setflag(3); # ... even if the transaction fails
                        }
                        route(RELAY);
                } else {
                        if (is_method("SUBSCRIBE") && uri == myself) {
                                # in-dialog subscribe requests
                                route(PRESENCE);
                                exit;
                        }
                        if ( is_method("ACK") ) {
                                if ( t_check_trans() ) {
                                        # non loose-route, but stateful ACK;
                                        # must be ACK after a 487 or e.g. 404 from upstream server
                                        t_relay();
                                        exit;
                                } else {
                                        # ACK without matching transaction ... ignore and discard.
                                        exit;
                                }
                        }
                        sl_send_reply("404","Not here");
                }
                exit;
        }
}

# Handle SIP registrations
route[REGISTRAR] {
        if(!is_method("REGISTER"))
                return;
        #sl_send_reply("404", "No registrar");
        #t_relay();
        if(!ds_select_dst("1", "4"))
        {
                sl_send_reply("404", "No registrar");
                exit;
        }
        forward();

        exit;
}

# Presence server route
route[PRESENCE] {
        if(!is_method("PUBLISH|SUBSCRIBE"))
                return;

        sl_send_reply("404", "Not here");
        exit;
}

# Dispatch requests
route[DISPATCH] {
        # round robin dispatching on gateways group '1'
        if(!ds_select_dst("1", "4"))
        {
                send_reply("404", "No destination");
                exit;
        }
        xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
        t_on_failure("RTF_DISPATCH");
        return;
}

# Sample failure route
failure_route[RTF_DISPATCH] {
        if (t_is_canceled()) {
                exit;
        }
        # next DST - only for 500 or local timeout
        if (t_check_status("500")
                        or (t_branch_timeout() and !t_branch_replied()))
        {
                if(ds_next_dst())
                {
                        t_on_failure("RTF_DISPATCH");
                        route(RELAY);
                        exit;
                }
        }
}

If I connect my asterisk box directly to my SIP provider it works perfectly. But if I connect it to the kamailio server and the kamailio server to the SIP provider it doesn't.

I googled for hours and tried a lot of things and I really have no idea what I could try next... If anyone could help me I would be very happy!

Thank you very much and best regards

Answer 1

I added my SIP provider credentials like this:

kamctl add test testpasswd - This is wrong.

Check following link for detailed info how you should set-up SIP trunk on Kamailio, which are using username/password authentication:

http://lists.sip-router.org/pipermail/sr-users/2015-September/090001.html