7 May
2015
7 May
'15
5:45 p.m.
THIS IS AN AUTOMATED MESSAGE, DO NOT REPLY.
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(a)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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