Kamailio SIP proxy for freeswitch with RTPEngine for RTP traffic (Kamailio+SIP+RTP+Freeswitch)

kamailio.cfg

#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v5.2 - default configuration script
#     - web: https://www.kamailio.org
#     - git: https://github.com/kamailio/kamailio
#
# Direct your questions about this file to: <sr-users@lists.kamailio.org>
#
# Refer to the Core CookBook at https://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#

####### Defined Values #########

#!define MULTIDOMAIN 0

# - flags
#   FLT_ - per transaction (message) flags
#   FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5

#!define FLB_NATB 6
#!define FLB_NATSIPPING 7

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

### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
debug=2
log_stderror=no

memdbg=5
memlog=5

log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "

/* number of SIP routing processes */
children=2

/* uncomment the next line to disable TCP (default on) */
# disable_tcp=yes

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

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

/* listen addresses */
# listen=udp:127.0.0.1:5060


####### Custom Parameters #########

/* These parameters can be modified runtime via RPC interface
 * - see the documentation of 'cfg_rpc' module.
 *
 * Format: group.id = value 'desc' description
 * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */

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

/* set paths to location of modules */
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.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 "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
loadmodule "path.so"
loadmodule "rtpengine.so"

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


# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
# modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
# modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock")

# ----- ctl params -----
/* set the path to RPC unix socket control file */
# modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")

# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)

# ----- rr params -----
# set next param to 1 to add value to ;lr param (helps with some 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 -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
 * if you enable this parameter, be sure the enable "append_fromtag"
 * in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra",
    "src_user=$fU;src_domain=$fd;src_ip=$si;"
    "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)

modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223")

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


/* Main SIP request routing logic
 * - processing of any incoming SIP request starts with this route
 * - note: this is the same as route { ... } */
request_route {
    # RTP route
    if (is_method("INVITE")) {
        if (has_body("application/sdp")) {
            if (rtpengine_offer())
                t_on_reply("1");
        } else {
            t_on_reply("2");
        }
    }

    if (is_method("ACK") && has_body("application/sdp")) {
        rtpengine_answer();
    }

    # per request initial checks
    route(REQINIT);

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

    # handle retransmissions
    if (!is_method("ACK")) {
        if(t_precheck_trans()) {
            t_check_trans();
            exit;
        }
        t_check_trans();
    }

    # handle requests within SIP dialogs
    route(WITHINDLG);

    ### only initial requests (no To tag)

    # 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();
    }

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

    # Add path for REGISTER
    if ((is_method("REGISTER"))) {
        add_path();
    }

    # update $du to set the destination address for proxying
    $du = "sip:" + $rd + ":5060";
    route(RELAY);
    exit;
}

onreply_route[1]
{
    if (has_body("application/sdp"))
        rtpengine_answer();
}

onreply_route[2]
{
    if (has_body("application/sdp"))
        rtpengine_offer();
}

# Wrapper for relaying requests
route[RELAY] {

    # enable additional event routes for forwarded requests
    # - serial forking, RTP relaying handling, a.s.o.
    if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
        if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
    }
    if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
        if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
    }
    if (is_method("INVITE")) {
        if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
    }

    if (!t_relay()) {
        sl_reply_error();
    }
    exit;
}

# Per SIP request initial checks
route[REQINIT] {
    if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
        # silent drop for scanners - uncomment next line if want to reply
        # sl_send_reply("200", "OK");
        exit;
    }

    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    }

    if(is_method("OPTIONS") && uri==myself && $rU==$null) {
        sl_send_reply("200","Keepalive");
        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()) return;

    # sequential request withing a dialog should
    # take the path determined by record-routing
    if (loose_route()) {
        if (is_method("BYE")) {
            setflag(FLT_ACC); # do accounting ...
            setflag(FLT_ACCFAILED); # ... even if the transaction fails
        } else if ( is_method("NOTIFY") ) {
            # Add Record-Route for in-dialog NOTIFY as per RFC 6665.
            record_route();
        }
        route(RELAY);
        exit;
    }

    if ( is_method("ACK") ) {
        if ( t_check_trans() ) {
            # no loose-route, but stateful ACK;
            # must be an ACK after a 487
            # or e.g. 404 from upstream server
            route(RELAY);
            exit;
        } else {
            # ACK without matching transaction ... ignore and discard
            exit;
        }
    }
    sl_send_reply("404","Not here");
    exit;
}

# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
    xdbg("new branch [$T_branch_idx] to $ru\n");
}

# Manage incoming replies
onreply_route[MANAGE_REPLY] {
    xdbg("incoming reply\n");
}

# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
    if (t_is_canceled()) exit;
}