[tracker] Task opened: Presence blf server memory leak - sr-dev

7 May 2015


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A new Flyspray task has been opened.  Details are below.
User who did this - David Kovarik (kovik)
Attached to Project - sip-router
Summary - Presence blf server memory leak
Task Type - Bug Report
Category - Module
Status - Unconfirmed
Assigned To - 
Operating System - Linux
Severity - Low
Priority - Normal
Reported Version - Development
Due in Version - Undecided
Due Date - Undecided
Details - ======Kamailio presence server never release memory.======
I create test enviromet on one openvz server where run:
* kamailio stable 4.2.4 (x86_64/linux) 97f1ee, as registrar and main proxy - only relay SIPP uas->uac calls
  * kamailio develop 4.3.0-pre1 (x86_64/linux) 451fbe, as presence server
  * sipp uas and uac
  * connect sip phone grandstream GXP2160 with 24 blf buttons
=====TESTS:=====
<code>
free memory on start:
             total       used       free     shared    buffers     cached
Mem:       2800000     942080    1857920          0          0     170008
-/+ buffers/cache:     772072    2027928
Swap:            0          0          0
----------------------------------------
Presence          Kbytes     RSS   Dirty
total kB          140188    6212    3496 pid:8033
total kB          140188    4180    1728 pid:8035
total kB          140188    2644    1428 pid:8036
total kB          140188    4184    1732 pid:8037
total kB          140188    2652    1436 pid:8038
total kB          140188    2024    1232 pid:8039
total kB          140188    2640    1840 pid:8040
total kB          140192    1832    1204 pid:8041
total kB          140188    2116    1204 pid:8042
total kB          140188    2160    1544 pid:8043
----------------------------------------
Main proxy        Kbytes     RSS   Dirty
total kB          156240    7456    4000 pid:8062
total kB          156240    5076    2228 pid:8066
total kB          156240    4860    2028 pid:8067
total kB          156240    5232    2236 pid:8068
total kB          156240    4828    2012 pid:8069
total kB          156240    2916    1736 pid:8070
total kB          156240    3496    2456 pid:8071
total kB          156244    2280    1616 pid:8072
total kB          156240    2676    1632 pid:8073
total kB          156240    2596    1624 pid:8074
total kB          156240    2776    1684 pid:8075
----------------------------------------
make 10000 calls with sipp
----------------------------- Statistics Screen ------- [1-9]: Change Screen --
  Start Time             | 2015-05-06   14:11:11.983305 1430914271.983305
  Last Reset Time        | 2015-05-06   16:58:11.995418 1430924291.995418
  Current Time           | 2015-05-06   16:58:11.995471 1430924291.995471
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:000000           | 02:47:00:012000
  Call Rate              |    0.000 cps              |    0.998 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        0
  OutGoing call created  |        0                  |    10000
  Total Call created     |                           |    10000
  Current Call           |        0                  |
-------------------------+---------------------------+--------------------------
  Successful call        |        0                  |     9991
  Failed call            |        0                  |        9
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:000000           | 00:00:01:012000
  Call Length            | 00:00:00:000000           | 00:00:02:019000
------------------------------ Test Terminated --------------------------------
memory after 10000 calls:
/root/count_mem.sh
             total       used       free     shared    buffers     cached
Mem:       2800000    2637760     162240          0          0     178412
-/+ buffers/cache:    2459348     340652
Swap:            0          0          0
----------------------------------------
Presence          Kbytes     RSS   Dirty
total kB          140188    6212    3496 pid:8033
total kB          561788  431240  428300 pid:8035
total kB          564800  434212  431288 pid:8036
total kB          565200  434596  431656 pid:8037
total kB          564576  433952  431028 pid:8038
total kB          140188    8640    6784 pid:8039
total kB          140188    7184    6328 pid:8040
total kB          140192    1832    1204 pid:8041
total kB          140188    2360    1272 pid:8042
total kB          140188    2160    1544 pid:8043
----------------------------------------
Main proxy        Kbytes     RSS   Dirty
total kB          156240    7456    4000 pid:8062
total kB          156240   13984    9996 pid:8066
total kB          156240   13956    9996 pid:8067
total kB          156240   13980    9996 pid:8068
total kB          156240   13796    9988 pid:8069
total kB          156240    6500    5128 pid:8070
total kB          156240   11064    9880 pid:8071
total kB          156244    2280    1616 pid:8072
total kB          156240    2676    1632 pid:8073
total kB          156240    3084    2088 pid:8074
total kB          156240    2776    1684 pid:8075
----------------------------------------
and next day morning:
 /root/count_mem.sh 
             total       used       free     shared    buffers     cached
Mem:       2800000    2637760     162240          0          0      39604
-/+ buffers/cache:    2598156     201844
Swap:            0          0          0
----------------------------------------
Presence          Kbytes     RSS   Dirty
total kB          140188    4936    3496 pid:8033
total kB          561788  430912  428300 pid:8035
total kB          564800  433900  431288 pid:8036
total kB          565200  434272  431660 pid:8037
total kB          564576  433644  431032 pid:8038
total kB          140188    8336    6788 pid:8039
total kB          140188    6880    6336 pid:8040
total kB          140192    1604    1204 pid:8041
total kB          140188    1936    1272 pid:8042
total kB          140188    1872    1544 pid:8043
----------------------------------------
Main proxy        Kbytes     RSS   Dirty
total kB          156240    5936    4000 pid:8062
total kB          156240   13644    9996 pid:8066
total kB          156240   13580    9996 pid:8067
total kB          156240   13600    9996 pid:8068
total kB          156240   13580    9996 pid:8069
total kB          156240    7064    5256 pid:8070
total kB          156240   10696    9880 pid:8071
total kB          156244    2008    1616 pid:8072
total kB          156240    2284    1632 pid:8073
total kB          156240    2736    2088 pid:8074
total kB          156240    2436    1684 pid:8075
----------------------------------------
</code>
=====Configs=====
====/root/count_mem.sh====
<code>
#!/bin/bash
free
echo "----------------------------------------"
echo "Presence          Kbytes     RSS   Dirty"
#pmap -x $(cat /var/run/kamailio_dev/kamailio_dev.pid)|grep total
for line in $(pgrep -f '/usr/local/kamailio-dev/sbin/kamailio -f /usr/local/kamailio-dev/etc/kamailio/kamailio.cfg -P /var/run/kamailio_dev/kamailio_dev.pid -m 16 -M 8 -u root -g root')
do
  pms=$(pmap -x $line|grep total)
  echo "$pms pid:$line"
done
echo "----------------------------------------"
echo "Main proxy        Kbytes     RSS   Dirty"
for line in $(pgrep -f '/usr/local/kamailio-4.2/sbin/kamailio -f /usr/local/kamailio-4.2/etc/kamailio/kamailio.cfg -P /var/run/kamailio_stable/kamailio_stable.pid -m 16 -M 8 -u root -g root')
do
  pms=$(pmap -x $line|grep total)
  echo "$pms pid:$line"
done
echo "----------------------------------------"
</code>
====Kamailio main server changes from stock config mark as #change====
<code>
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v4.2 - default configuration script
#     - web: http://www.kamailio.org
#     - git: http://sip-router.org
#
# Direct your questions about this file to: sr-users@lists.sip-router.org
#
# Refer to the Core CookBook at http://www.kamailio.org/wiki/
# 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
#
# *** To enable mysql: 
#!define WITH_MYSQL
#
# *** To enable authentication execute:
#     - enable mysql
#!define WITH_AUTH
#     - add users using 'kamctl'
#
# *** To enable IP authentication execute:
#     - enable mysql
#     - enable authentication
#     - define WITH_IPAUTH
#     - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
#     - enable mysql
#!define WITH_USRLOCDB
#
# *** To enable presence server execute:
#     - enable mysql
#!define WITH_PRESENCE
#
# *** To enable nat traversal execute:
#!define WITH_NAT
#     - install RTPProxy: http://www.rtpproxy.org
#     - start RTPProxy:
#        rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#     - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING
#
# *** To enable PSTN gateway routing execute:
#     - define WITH_PSTN
#     - set the value of pstn.gw_ip
#     - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
#     - enable mysql
#     - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
#     - enable mysql
#     - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
#     - enable mysql
#     - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
#     - adjust CFGDIR/tls.cfg as needed
#     - define WITH_TLS
#
# *** To enable XMLRPC support execute:
#     - define WITH_XMLRPC
#     - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
#     - adjust pike and htable=>ipban settings as needed (default is
#       block if more than 16 requests in 2 seconds and ban for 300 seconds)
#     - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
#     - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
#     - define WITH_VOICEMAIL
#     - set the value of voicemail.srv_ip
#     - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
#     - enable mysql
#     - define WITH_ACCDB
#     - add following columns to database
#!ifdef ACCDB_COMMENT
  ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#!endif
####### Include Local Config If Exists #########
import_file "kamailio-local.cfg"
####### Defined Values #########
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#       as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#change
#!define DBURL "mysql://root:root@localhost/stable"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
# - 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
#change
#!define FLT_PUBLISH 19
#!define FLT_DIALOG 18
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
####### Global Parameters #########
### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
#!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
/* 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"
/* 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:81.31.45.53:6688 #change
/* port to listen to
 * - can be specified more than once if needed to listen on many ports */
#port=6688
#!ifdef WITH_TLS
enable_tls=yes
#!endif
# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605
####### 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
#
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "PSTN GW Address"
pstn.gw_port = "" desc "PSTN GW Port"
#!endif
#!ifdef WITH_VOICEMAIL
# VoiceMail Routing on offline, busy or no answer
#
# - by default Voicemail server IP is empty to avoid misrouting
voicemail.srv_ip = "" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif
####### Modules Section ########
# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules/"
#!else
mpath="/usr/local/kamailio-4.2/lib64/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
loadmodule "mi_fifo.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 "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
#!ifdef WITH_PRESENCE
#change
loadmodule "dialog.so"
loadmodule "pua.so"
loadmodule "pua_dialoginfo.so"
#loadmodule "presence.so"
#loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
# ----------------- setting module-specific parameters ---------------
modparam("xlog", "buf_size", 8192)
modparam("xlog", "force_color", 0)
modparam("xlog", "prefix", "")
modparam("xlog", "long_format", 0)
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_stable_fifo")
# ----- 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", 0)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 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=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
        "src_user=$fU;src_domain=$fd;src_ip=$si;"
        "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "user_column", "username")
modparam("auth_db", "use_domain", MULTIDOMAIN)
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
#!endif
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- speeddial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
#!ifdef WITH_PRESENCE
# ----- presence params -----
#modparam("presence", "db_url", DBURL)
# ----- presence_xml params -----
#change
#modparam("presence_xml", "db_url", DBURL)
#modparam("presence_xml", "force_active", 1)
modparam("dialog", "dlg_flag",FLT_DIALOG)
modparam("dialog", "default_timeout",10800)
modparam("pua", "db_url", DBURL)
modparam("pua", "dlginfo_increase_version", 1)
modparam("pua", "outbound_proxy", "sip:81.31.45.53:9988")
modparam("pua_dialoginfo", "send_publish_flag", FLT_PUBLISH)
modparam("pua_dialoginfo", "use_pubruri_avps", 1)
modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:pcaller)")
modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:pcallee)")
modparam("pua_dialoginfo", "include_callid", 1)
modparam("pua_dialoginfo", "include_tags", 1)
modparam("pua_dialoginfo", "include_localremote", 1)
modparam("pua_dialoginfo", "override_lifetime", 300)
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/kamailio-4.2/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
modparam("debugger", "log_level_name", "exec")
#!endif
####### 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 {
# per request initial checks
        route(REQINIT);
# NAT detection
        route(NATDETECT);
# CANCEL processing
        if (is_method("CANCEL")) {
                if (t_check_trans()) {
                        route(RELAY);
                }
                exit;
        }
# handle requests within SIP dialogs
        route(WITHINDLG);
### only initial requests (no To tag)
# handle retransmissions
        if(t_precheck_trans()) {
                t_check_trans();
                exit;
        }
        t_check_trans();
# authentication
  #xlogl("L_INFO","PRISLO $rm z $si,$sp");
#change
  if($si=="81.31.45.53" && $sp=='5099'){
    $var(sipp)=1;
  }else{
          route(AUTH);
  }
# record routing for dialog forming requests (in case they are routed)
        # - remove preloaded route headers
        remove_hf("Route");
        if (is_method("INVITE|SUBSCRIBE"))
                record_route();
# account only INVITEs
        if (is_method("INVITE")) {
                setflag(FLT_ACC); # do accounting
        }
# dispatch requests to foreign domains
        route(SIPOUT);
### requests for my local domains
# handle presence related requests
        route(PRESENCE);
# handle registrations
        route(REGISTRAR);
if ($rU==$null) {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }
# dispatch destinations to PSTN
        route(PSTN);
# user location service
#change
  if($var(sipp)==1){
        route(RELAY);
  }else{
          route(LOCATION);
  }
}
# Wrapper for relaying requests
route[RELAY] {
#change
  if($var(sipp)==1){
    $du=$null;
    $var(sipp)=$null;
    $rd="81.31.45.53";
    $rp=5090;
  }
#xlogl("L_INFO","START RELAY $rm -- $ci");
        # 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");
$avp(s:pcaller)="sip:"+$fU+"@81.31.45.53:9988";
    $avp(s:pcallee)="sip:"+$rU+"@81.31.45.53:9988";
    setflag(FLT_PUBLISH);
    dlg_manage();
    #xlogl("L_INFO","START DIALOG - $rm -- $ci -- $avp(s:pcaller)->$avp(s:pcallee)");
}
if (!t_relay()) {
                sl_reply_error();
        }
        exit;
}
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
        # flood dection from same IP and traffic ban for a while
        # be sure you exclude checking trusted peers, such as pstn gateways
        # - local host excluded (e.g., loop to self)
        if(src_ip!=myself) {
                if($sht(ipban=>$si)!=$null) {
                        # ip is already blocked
                        xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
                        exit;
                }
                if (!pike_check_req()) {
                        xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
                        $sht(ipban=>$si) = 1;
                        exit;
                }
        }
        if($ua =~ "friendly-scanner") {
                sl_send_reply("200", "OK");
                exit;
        }
#!endif
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()) {
                route(DLGURI);
                if (is_method("BYE")) {
                        setflag(FLT_ACC); # do accounting ...
                        setflag(FLT_ACCFAILED); # ... even if the transaction fails
                }
                else if ( is_method("ACK") ) {
                        # ACK is forwarded statelessy
                        route(NATMANAGE);
                }
                else if ( is_method("NOTIFY") ) {
                        # Add Record-Route for in-dialog NOTIFY as per RFC 6665.
      #xlogl("L_INFO","IN DIALOG $rm -- $ci");
                        record_route();
                }
                route(RELAY);
                exit;
        }
if (is_method("SUBSCRIBE") && uri == myself) {
                # in-dialog subscribe requests
                route(PRESENCE);
                exit;
        }
        if ( is_method("ACK") ) {
                if ( t_check_trans() ) {
                        # no loose-route, but stateful ACK;
                        # must be an ACK after a 487
                        # or e.g. 404 from upstream server
                        route(RELAY);
                        exit;
                } else {
                        # ACK without matching transaction ... ignore and discard
                        exit;
                }
        }
        sl_send_reply("404","Not here");
        exit;
}
# Handle SIP registrations
route[REGISTRAR] {
        if (!is_method("REGISTER")) return;
if(isflagset(FLT_NATS)) {
                setbflag(FLB_NATB);
#!ifdef WITH_NATSIPPING
                # do SIP NAT pinging
                setbflag(FLB_NATSIPPING);
#!endif
        }
        if (!save("location"))
                sl_reply_error();
        exit;
}
# User location service
route[LOCATION] {
#!ifdef WITH_SPEEDDIAL
        # search for short dialing - 2-digit extension
        if($rU=~"^[0-9][0-9]$")
                if(sd_lookup("speed_dial"))
                        route(SIPOUT);
#!endif
#!ifdef WITH_ALIASDB
        # search in DB-based aliases
        if(alias_db_lookup("dbaliases"))
                route(SIPOUT);
#!endif
$avp(oexten) = $rU;
        if (!lookup("location")) {
                $var(rc) = $rc;
                route(TOVOICEMAIL);
                t_newtran();
                switch ($var(rc)) {
                        case -1:
                        case -3:
                                send_reply("404", "Not Found");
                                exit;
                        case -2:
                                send_reply("405", "Method Not Allowed");
                                exit;
                }
        }
# when routing via usrloc, log the missed calls also
        if (is_method("INVITE")) {
                setflag(FLT_ACCMISSED);
        }
route(RELAY);
        exit;
}
# Presence server processing
route[PRESENCE] {
        if(!is_method("PUBLISH|SUBSCRIBE"))
                return;
#       if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") {
#               route(TOVOICEMAIL);
                # returns here if no voicemail server is configured
#               sl_send_reply("404", "No voicemail service");
#               exit;
#       }
#!ifdef WITH_PRESENCE
        if (!t_newtran()) {
                sl_reply_error();
                exit;
        }
#preposlu na presence server
  $rd="81.31.45.53";
  $rp=9988;
  if(!t_relay_to_udp()){
    sl_reply_error();
  }
#       if(is_method("PUBLISH")) {
#               handle_publish();
#               t_release();
#       } else if(is_method("SUBSCRIBE")) {
#               handle_subscribe();
#               t_release();
#       }
        exit;
#!endif
# if presence enabled, this part will not be executed
        if (is_method("PUBLISH") || $rU==$null) {
                sl_send_reply("404", "Not here");
                exit;
        }
        return;
}
# IP authorization and user uthentication
route[AUTH] {
#!ifdef WITH_AUTH
#!ifdef WITH_IPAUTH
        if((!is_method("REGISTER")) && allow_source_address()) {
                # source IP allowed
                return;
        }
#!endif
if (is_method("REGISTER") || from_uri==myself)
        {
                # authenticate requests
                if (!auth_check("$fd", "subscriber", "1")) {
                        auth_challenge("$fd", "0");
                        exit;
                }
                # user authenticated - remove auth header
                if(!is_method("REGISTER|PUBLISH"))
                        consume_credentials();
        }
        # if caller is not local subscriber, then check if it calls
        # a local destination, otherwise deny, not an open relay here
        if (from_uri!=myself && uri!=myself) {
                sl_send_reply("403","Not relaying");
                exit;
        }
#!endif
        return;
}
# Caller NAT detection
route[NATDETECT] {
#!ifdef WITH_NAT
        force_rport();
        if (nat_uac_test("19")) {
                if (is_method("REGISTER")) {
                        fix_nated_register();
                } else {
                        if(is_first_hop())
                                set_contact_alias();
                }
                setflag(FLT_NATS);
        }
#!endif
        return;
}
# RTPProxy control and singaling updates for NAT traversal
route[NATMANAGE] {
#!ifdef WITH_NAT
        if (is_request()) {
                if(has_totag()) {
                        if(check_route_param("nat=yes")) {
                                setbflag(FLB_NATB);
                        }
                }
        }
        if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
                return;
rtpproxy_manage("co");
if (is_request()) {
                if (!has_totag()) {
                        if(t_is_branch_route()) {
                                add_rr_param(";nat=yes");
                        }
                }
        }
        if (is_reply()) {
                if(isbflagset(FLB_NATB)) {
                        if(is_first_hop())
                                set_contact_alias();
                }
        }
#!endif
        return;
}
# URI update for dialog requests
route[DLGURI] {
#!ifdef WITH_NAT
        if(!isdsturiset()) {
                handle_ruri_alias();
        }
#!endif
        return;
}
# Routing to foreign domains
route[SIPOUT] {
        if (uri==myself) return;
append_hf("P-hint: outbound\r\n");
        route(RELAY);
        exit;
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
        # check if PSTN GW IP is defined
        if (strempty($sel(cfg_get.pstn.gw_ip))) {
                xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
                return;
        }
# route to PSTN dialed numbers starting with '+' or '00'
        #     (international format)
        # - update the condition to match your dialing rules for PSTN routing
        if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$"))
                return;
# only local users allowed to call
        if(from_uri!=myself) {
                sl_send_reply("403", "Not Allowed");
                exit;
        }
if (strempty($sel(cfg_get.pstn.gw_port))) {
                $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
        } else {
                $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
                                        + $sel(cfg_get.pstn.gw_port);
        }
route(RELAY);
        exit;
#!endif
return;
}
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
        # allow XMLRPC from localhost
        if ((method=="POST" || method=="GET")
                        && (src_ip==127.0.0.1)) {
                # close connection only for xmlrpclib user agents (there is a bug in
                # xmlrpclib: it waits for EOF before interpreting the response).
                if ($hdr(User-Agent) =~ "xmlrpclib")
                        set_reply_close();
                set_reply_no_connect();
                dispatch_rpc();
                exit;
        }
        send_reply("403", "Forbidden");
        exit;
}
#!endif
# Routing to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
        if(!is_method("INVITE|SUBSCRIBE"))
                return;
# check if VoiceMail server IP is defined
        if (strempty($sel(cfg_get.voicemail.srv_ip))) {
                xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
                return;
        }
        if(is_method("INVITE")) {
                if($avp(oexten)==$null)
                        return;
                $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
                                + ":" + $sel(cfg_get.voicemail.srv_port);
        } else {
                if($rU==$null)
                        return;
                $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)
                                + ":" + $sel(cfg_get.voicemail.srv_port);
        }
        route(RELAY);
        exit;
#!endif
return;
}
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
        xdbg("new branch [$T_branch_idx] to $ru\n");
        route(NATMANAGE);
}
# Manage incoming replies
onreply_route[MANAGE_REPLY] {
        xdbg("incoming reply\n");
        if(status=~"[12][0-9][0-9]")
                route(NATMANAGE);
}
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
        route(NATMANAGE);
if (t_is_canceled()) {
                exit;
        }
#!ifdef WITH_BLOCK3XX
        # block call redirect based on 3xx replies.
        if (t_check_status("3[0-9][0-9]")) {
                t_reply("404","Not found");
                exit;
        }
#!endif
#!ifdef WITH_VOICEMAIL
        # serial forking
        # - route to voicemail on busy or no answer (timeout)
        if (t_check_status("486|408")) {
                $du = $null;
                route(TOVOICEMAIL);
                exit;
        }
#!endif
}
</code>
====Kamailio presence server====
<code>
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v4.3 - default configuration script
#     - web: http://www.kamailio.org
#     - git: http://sip-router.org
#
#!define WITH_MYSQL
#!define WITH_PRESENCE
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#       as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#!define DBURL "mysql://root:root@localhost/presence"
#!endif
#!endif
####### Global Parameters #########
### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
#!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
listen=udp:81.31.45.53:9988
mpath="/usr/local/kamailio-dev/lib64/kamailio/modules/"
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
loadmodule "mi_fifo.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 "mi_rpc.so"
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
loadmodule "presence_dialoginfo.so"
#!endif
modparam("xlog", "buf_size", 8192)
modparam("xlog", "force_color", 0)
modparam("xlog", "prefix", "")
modparam("xlog", "long_format", 0)
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/var/run/kamailio_dev/kamailio_fifo")
# ----- ctl params -----
modparam("ctl", "binrpc", "unix:/var/run/kamailio_dev/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", 0)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
modparam("presence", "server_address", "sip:81.31.45.53:9988")
#modparam("presence", "enable_sphere_check", 1)
modparam("presence", "local_log_level", 3)
#modparam("presence", "subs_remove_match", 1)
#modparam("presence", "retrieve_order", 1)
#modparam("presence", "sip_uri_match", 1)
#modparam("presence", "subs_db_mode", 0)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
modparam("presence_dialoginfo", "force_single_dialog", 1) 
modparam("presence_dialoginfo", "force_dummy_dialog", 1)
#!endif
####### 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 {
# per request initial checks
        route(REQINIT);
# handle presence related requests
        route(PRESENCE);
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;
        }
}
# Presence server processing
route[PRESENCE] {
        if(!is_method("PUBLISH|SUBSCRIBE"))
                return;
if (!t_newtran()) {
                sl_reply_error();
                exit;
        }
#xlogl("L_INFO","PRES $ci $rm");
        if(is_method("PUBLISH")) {
                handle_publish();
                t_release();
        } else if(is_method("SUBSCRIBE")) {
                handle_subscribe();
                t_release();
        }
        exit;
}
</code>
====SIPP UAS====
sipp -sf uas.xml 81.31.45.53 -p 5090 -trace_err
<code>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'uas' scenario.                       -->
<!--                                                                    -->
<scenario name="Basic UAS responder">
  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  <!-- are saved and used for following messages sent. Useful to test   -->
  <!-- against stateful SIP proxies/B2BUAs.                             -->
  <recv request="INVITE" crlf="true" rrs="true">
  </recv>
<!-- The '[last_*]' keyword is replaced automatically by the          -->
  <!-- specified header if it was present in the last message received  -->
  <!-- (except if it was a retransmission). If the header was not       -->
  <!-- present or if no message has been received, the '[last_*]'       -->
  <!-- keyword is discarded, and all bytes until the end of the line    -->
  <!-- are also discarded.                                              -->
  <!--                                                                  -->
  <!-- If the specified header was present several times in the         -->
  <!-- message, all occurences are concatenated (CRLF separated)        -->
  <!-- to be used in place of the '[last_*]' keyword.            Record-[routes]       -->
<send rrs="true">
    <![CDATA[
SIP/2.0 180 Ringing
      [last_Via:]
      Record-[routes]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:[call_number]@[local_ip]:[local_port];transport=[transport]
      Content-Length: 0
]]>
  </send>
<pause milliseconds = "1000"/>
  <!--
  -->
  <send retrans="500" rrs="true">
    <![CDATA[
SIP/2.0 200 OK
      [last_Via:]
      Record-[routes]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:[call_number]@[local_ip]:[local_port];transport=[transport]
      Content-Type: application/sdp
      Content-Length: [len]
v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000
]]>
  </send>
<!--
  <nop>
    <action>
      <exec play_pcap_audio="/root/sipp/g711a.pcap"/>
    </action>
  </nop>
  -->
  <recv request="ACK"
        optional="true"
        rtd="true"
        crlf="true"
        rrs="true">
  </recv>
<recv request="BYE" rrs="true">
  </recv>
<send rrs="true">
    <![CDATA[
SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:[call_number]@[local_ip]:[local_port];transport=[transport]
      Content-Length: 0
]]>
  </send>
<!-- Keep the call open for a while in case the 200 is lost to be     -->
  <!-- able to retransmit it if we receive the BYE again.
<timewait milliseconds="1000"/>
<timewait milliseconds="500"/>
  -->
  <!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
</code>
====SIPP UAC====
sipp -sf ./uac.xml -p 5099 -s uas -r 1 -rp 1000 -m 10000 81.31.45.53:6688 -trace_err -inf ./inf3.csv
uac.xml
<code>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'uac' scenario.                       -->
<!--                                                                    -->
<scenario name="Basic Sipstone UAC">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.        Diversion: <tel:0042910116106>        -->
  <send retrans="500">
    <![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field] sip:[field0]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
      To: [field1] sip:[field1]@[remote_ip]:[remote_port]
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Content-Type: application/sdp
      Content-Length: [len]
v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000
]]>
  </send>
<recv response="100" optional="true">
  </recv>
<recv response="180" optional="true" rrs="true">
  </recv>
<recv response="183" optional="true" rrs="true">
  </recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  <!-- are saved and used for following messages sent. Useful to test   -->
  <!-- against stateful SIP proxies/B2BUAs.                             -->
  <recv response="200" rtd="true" rrs="true">
  </recv>
<!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.    [routes]   -->
  <!-- Contact: sip:sipp@[local_ip]:[local_port] sip:[field1]@[remote_ip]:[remote_port] -->
  <send rrs="true">
    <![CDATA[
ACK [next_url] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [routes]
      From: [field0] sip:[field0]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
      To: [field1] sip:[field1]@[remote_ip]:[remote_port][peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Content-Length: 0
]]>
  </send>
<pause milliseconds = "1000"/>
  <!--
  -->
  <!-- This delay can be customized by the -d command-line option       -->
  <!-- or by adding a 'milliseconds = "value"' option here.             -->
<!-- The 'crlf' option inserts a blank line in the statistics report. [routes]  -->
  <send retrans="500" rrs="true">
    <![CDATA[
BYE [next_url] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [routes]
      From: [field0] sip:[field0]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
      To: [field1] sip:[field1]@[remote_ip]:[remote_port][peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 BYE
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0
]]>
  </send>
<recv response="200" crlf="true" rrs="true">
  </recv>
<!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
</code>
====inf3.csv====
<code>
SEQUENTIAL
1;2
2;3
3;4
4;5
5;6
6;7
7;8
8;9
9;10
10;11
11;12
12;24
24;23
23;22
22;21
21;20
20;19
19;18
18;17
17;16
16;15
15;14
14;13
13;1
</code>
On grandstream gxp216, sets programable keys to blf 1-24.
BLF works correctly, only problem is a memory leak.
The stable kamailio as presence server leak too.
Thank you.
More information can be found at the following URL:
http://sip-router.org/tracker/index.php?do=details&task_id=507
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