Telephony Solutions with Kamailio

  • Features
    • RFC3261 compliant
    • Customisable and Felxible – Modular Architecture
    • Call routing and control
    • NAT traversal
    • Data store integration
    • Transport Layers supported
    • Asynchronous TCP, UDP and SCTP
    • Secure Communication ( TLS + AAA)
    • IP and DNS
    • Accounting
    • External Integration
    • Rich Communication Services ( RCS)
  • Kamailio (OpenSER) SIP Server v4.3- default configuration script
    • debug
    • authentication
    • persistent user location
    • presence
    • nat traversal
    • RTPProxy
    • PSTN gateway
    • Block 3XX redirect repliest
    • Database aliases lookup
    • speed dial
    • multi-domain support
    • XMLRPC
    • anti-flood detection
  • Routing Logic
    • WITHINDLG
    • LOCATION
    • PRESENCE
    • AUTH
    • NATDETECT
    • NATMANAGE
    • DLGURI
    • SIPOUT
    • PSTN
    • XMLRPC
    • TOVOICEMAIL
    • MANAGE_BRANCH
    • MANAGE_REPLY
    • MANAGE_FAILURE
  • kamctl
  • kamdbctl
  • Kamctlrc
  • Kamailio.cfg
  • kamcmd
  • siremis

Kamailio™ (former OpenSER) is an Open Source SIP Server released under GPL.

Kamailio primarily acts as a SIP server for VOIP and telecommunications platforms under various roles and can handle load of hight CPS ( Calls per second ) with custom call routing logic with the help of scripts.

Rich features set suiting to telephony domain that includes IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; Json and XMLRPC control interface, SNMP monitoring.

To integrate with a carrier grade telecom network as SBC / gateway / inbound/outbound proxy , it can act as IPv4-IPv6 gateway , UDP/TCP/SCTP/WS translator and even had NAT and anti DOS attack support .

Features

If Kamailio is the central to the VoIP system it can also perform accounting with rich database extensions Mysql PostgreSQL UnixODBC Berkeley DB Oracle Redis, MongoDB Cassandra etc

  • Kamailio is SIP (RFC3261) compliant

It can work as Registrar or Location server. For SIP call logic it can become a Proxy or SIP Application server. Can also act like a Redirect, Dispatcher or simply a SIP over websocket server.

  • Customisable and Felxible

It can be embedded to devices as the binary file is small size. Additional modules can be appended for more functions with the same core.

Modular architecture – core, internal libraries , module interface and ability to extend functionality with scripts such as LUA, Kamailio can be readily integrated to a VOIP ecosystem.

Kamailio Architecture , Core and Modules.

  • Call routing and control with Scripting and programming

Offers stateless and transactional stateful SIP Proxy processing ( suited for inbound gateways ) and serial and parallel forking.

Kamailio Transaction management describes branches, serial and paralle forking and TM module.

Also NAT traversal support for SIP and RTP traffic ( suited to be WebRTC server ) . Read more about kamailio DNS subsystem management , load balancing , NAT and NAThelper modules in Kamailio DNS and NAT.

Kamailio as a WebRTC SIP Server with mdules websocket , TLS , NATHelper  and JSSIP integration.

Among other features it offers load balancing with many distribution algorithms and failover support, flexible least cost routing , routing failover and replication for High Availability (HA).

Can be readily integrated with external databases, caches, notification system ( SNS , APNS , GCM ), VoIP monitors, CDR processors, API systems etc for  efficient call processing.

  • Transport Layers supported 
    • UDP, TCP, TLS and SCTP
    • IPv4 and IPv6
    • gateways via (IPv4 to IPv6, UDP to TLS, a.s.o.)
    • SCTP multi-homing and multi-streaming
    • WebSocket for WebRTC 
  • Asynchronous TCP, UDP and SCTP

Asynchronous SIP message processing and  inter-process message queues communication system

  • Secure Communication ( TLS  + AAA)
    • Digest SIP User authentication
    • Authorization via ACL or group membership
    • IP and Network authentication
    • TLS support for SIP signaling
    • transparent handling of SRTP for secure audio
    • TLS domain name extension support
    • authentication and authorization against database (MySQL, PostgreSQL, UnixODBC, BerkeleyDB, Oracle, text files), RADIUS and DIAMETER

Kamailio Security here for snaity, ACL permission , firewall , flood detection , topology hiding and digests.

  • IP and DNS
    • support for SRV and NAPTR DNS lookups
    • SRV DNS failover
    • DNSsec support
    • ENUM support
    • internal DNS caching system – avoid DNS blocking
    • IP level Blacklists
    • multi-homed and multi-domain support
    • topology hiding – hide IP addresses in SIP headers to protect your network architecture
  • Accounting

Kamailio gives event based and configurable accounting data details. Can show multi-leg call accounting ( A leg to B leg ). It can store to database, Radius or Diameter based on module used . Has a prepaid engine.

  • External Interaction

text-based management interface via FIFO file, udp, xmlrpc and unix sockets.

RPC control interface – via XMLRPC, UDP or TCP

  • Rich Communication Services (RCS)
    • SIP SIMPLE Presence Server (rich presence)
    • Presence User Agent ( SUBSCRIBE , NOTIFY and PUBLSH)
    • XCAP client capabilities and Embedded XCAP Server
    • Presence DialogInfo support – SLA/BLA
    • Instant Messaging ( IM) 
    • Embedded MSRP relay
  • Monitoring and Troubleshooting

Support for SNMP – interface to Simple Network Management Protocol.  For Debugging it has config debugger , remote control via XMLRPC and error message logging system .Provides internal statistics exported via RPC and SNMP.

  • Extensibility APIs

The supported  one are Perl  , Java SIP Servlet Application Interface  , Lua  , Managed Code (C#) , Python.

Lua Scripts for kamailio Routing – KEMI interpreter , function , routing logic, PV variables

  • Multiple Database Backends

(MySQL, PostgreSQL, SQLite, UnixODBC, BerkeleyDB, Oracle, text files) and other database types which have unixodbc drivers. ‘

It can have connections pool and different backends  be used at same time (e.g., accounting to Oracle and authorization against MySQL).

Has connectors for Memcached, Redis , MongoDB and Cassandra no-SQL backends

  • Interconnectivity

Acts as SIP to PSTN gateway and gateway to sms or xmpp and other IM services. Has Interoperability with SIP enabled devices and applications such as SIP phones (Snom, Cisco, etc.), Media Servers (Asterisk, FreeSwitch, etc). More details on Kamailio as Inbound/Outbound proxy or Session Border Controller (SBC) here

  • IMS
    • diameter support and authentication
    • I-CSCF, P-CSCF, S-CSCF
    • charging, QOS, ISC
  • Miscellaneous
    • CPL – Call Processing Language (RFC3880)
    • Internal generic caching system
    • Memcached connector
    • Redis NoSQL database connector
    • CLI – kamctl and sercmd
    • Web Management Interface: Siremis
    • SIP-T and SIP-I
    • music on hold queue
    • message body compression/decompression (gzip-deflate)

Scalability with Kamailio system

  • Kamailio can run on embedded systems, with limited resources – the performances can be up to hundreds of call setups per second
  • used as load balancer in stateless mode, Kamailio can handle over 5000 call setups per second
  • on systems with 4GB memory, Kamailio can serve a population over 300 000 online subscribers
  • system can easily scale by adding more Kamailio servers
  • Kamailio can be used in geographic distributed VoIP platforms
  • Kamailio least-cost-routing scales up to millions of routing rules
  • straightforward failover and redundancy

Start Kamalio

service kamailo start

Kamaiio Logs tailing

tail -f /var/log/kamailio

To Check if Kamailio instance is running

>ps -ax | grep “kamailio”
 57411 ?        S      0:01 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 4096 -M 128 -u root -g root

Read RTP engine on kamailio SIP server which focuses on setting up sipwise rtpegine to proxy rtp traffic from kamailio app server. Also daemon and kernal modules. ,transcoding , in-kernel packet forwarding , ngcontrol protocol etc.

Installation and Configuration

Installing kamailio from git repo

clone kamailio from their github https://github.com/kamailio/kamailio

goto desired branch. The contents of clonned folder are

COPYING    ChangeLog    INSTALL    ISSUES       Makefile        README README.md       doc       etc      misc       pkg      test      utils

run ‘make cfg’ which compiles using gcc abd created ‘src’ folder with contents

make cfg

Makefile Makefile.defs Makefile.groups Makefile.modules Makefile.rules Makefile.sources Makefile.utils core main.c modules.lst Makefile.cfg Makefile.dirs Makefile.libs Makefile.radius Makefile.shared Makefile.targets config.mak lib modules

Edit modules.lst to enable db_mysql and dialplan module

vim src/modules.lst

//the list of extra modules to compile
include_modules= db_mysql

Use ‘make’ command followed by make all. Make sure to have gcc bison and flex installed

make cfg
make all

Alternativey on debian system , use readymade measures like

apt install mysql-server
apt install kamailio kamailio-mysql-modules

To validate and verify the location of kamillio use ‘which kamailio’ which returns /usr/sbin/kamailio

For Modules installation, check all avaible modules with command ‘apt search kamailio’and to install a new module such as websockt module use ‘apt install kamailio-websocket-modules’

Database access : After installaing kamailio , edit the kamailio.cfg file in /etc/kamailio to set the reachabe SIP domain, database engine, username/password etc to connect to databaseand enable the kamdbctl script to run and create users and tables, etc.

SIP_DOMAIN=kamailio.org

SIP_DOMAIN=17.3.4.5

chrooted directory

$CHROOT_DIR=”/path/to/chrooted/directory”

database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE by default none is loaded

DBENGINE=MYSQL

Run kamdbctl to create users and database now

kamdbctl create

the database created is name kamailio and its tables are

+---------------------+
 | Tables_in_kamailio  |
 +---------------------+
 | acc                 |
 | acc_cdrs            |
 | active_watchers     |
 | address             |
 | aliases             |
 | carrier_name        |
 | carrierfailureroute |
 | carrierroute        |
 | cpl                 |
 | dbaliases           |
 | dialog              |
 | dialog_vars         |
 | dialplan            |
 | dispatcher          |
 | domain              |
 | domain_attrs        |
 | domain_name         |
 | domainpolicy        |
 | globalblacklist     |
 | grp                 |
 | htable              |
 | imc_members         |
 | imc_rooms           |
 | lcr_gw              |
 | lcr_rule            |
 | lcr_rule_target     |
 | location            |
 | location_attrs      |
 | missed_calls        |
 | mohqcalls           |
 | mohqueues           |
 | mtree               |
 | mtrees              |
 | pdt                 |
 | pl_pipes            |
 | presentity          |
 | pua                 |
 | purplemap           |
 | re_grp              |
 | rls_presentity      |
 | rls_watchers        |
 | rtpproxy            |
 | sca_subscriptions   |
 | silo                |
 | sip_trace           |
 | speed_dial          |
 | subscriber          |
 | trusted             |
 | uacreg              |
 | uid_credentials     |
 | uid_domain          |
 | uid_domain_attrs    |
 | uid_global_attrs    |
 | uid_uri             |
 | uid_uri_attrs       |
 | uid_user_attrs      |
 | uri                 |
 | userblacklist       |
 | usr_preferences     |
 | version             |
 | watchers            |
 | xcap                |
 +---------------------+

Kamctlrc

The Kamailio configuration file for the control tools. Can set variables used in the kamctl and kamdbctl setup scripts. Per default all variables here are commented out, the control tools will use their internal default values. This file lets to edit  SIP domain, the database engine, username/password/ to connect to database, etc.

## your SIP domain
 SIP_DOMAIN=1.1.1.1
## chrooted directory
# $CHROOT_DIR="/path/to/chrooted/directory"
## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE
# by default none is loaded

# If you want to setup a database with kamdbctl, you must at least specify this parameter.

DBENGINE=MYSQL
## database host
# DBHOST=localhost
# DBPORT=3306
## database name (for ORACLE this is TNS name)
# DBNAME=kamailio
# database path used by dbtext, db_berkeley or sqlite
# DB_PATH="/usr/local/etc/kamailio/dbtext"
database read/write user
# DBRWUSER="kamailio"
## password for database read/write user
# DBRWPW="kamailiorw"

database read only user

# DBROUSER="kamailioro"
## password for database read only user
# DBROPW="kamailioro"
## database access host (from where is kamctl used)
# DBACCESSHOST=192.168.0.1

database super user (for ORACLE this is ‘scheme-creator’ user)

# DBROOTUSER="root"
## password for database super user
## - important: this is insecure, targeting the use only for automatic testing
## - known to work for: mysql
# DBROOTPW="dbrootpw"
## database character set (used by MySQL when creating database)
#CHARSET="latin1"
## user name column
# USERCOL="username"
# SQL definitions

# If you change this definitions here, then you must change them
# in db/schema/entities.xml too.
# FIXME
# FOREVER="2030-05-28 21:32:15"
# DEFAULT_Q="1.0"
# Program to calculate a message-digest fingerprint
# MD5="md5sum"
# awk tool
# AWK="awk"
# gdb tool
# GDB="gdb"

# If you use a system with a grep and egrep that is not 100% gnu grep compatible,
# e.g. solaris, install the gnu grep (ggrep) and specify this below.

grep tool
# GREP="grep"
# egrep tool
# EGREP="egrep"
# sed tool
# SED="sed"
# tail tool
# LAST_LINE="tail -n 1"
# expr tool
# EXPR="expr"

Describe what additional tables to install. Valid values for the variables below are yes/no/ask. With ask (default) it will interactively ask the user for an answer, while yes/no allow for automated, unassisted installs.

#If to install tables for the modules in the EXTRA_MODULES variable.

# INSTALL_EXTRA_TABLES=ask
# If to install presence related tables.
# INSTALL_PRESENCE_TABLES=ask
# If to install uid modules related tables.
# INSTALL_DBUID_TABLES=ask

 Define what module tables should be installed.

If you use the postgres database and want to change the installed tables, then you must also adjust the STANDARD_TABLES or EXTRA_TABLES variable accordingly in the kamdbctl.base script.

standard modules

# STANDARD_MODULES="
standard acc lcr domain group permissions registrar usrloc msilo
alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher
dialplan"

extra modules

# EXTRA_MODULES="imc cpl siptrace domainpolicy carrierroute userblacklist htable purple sca"
type of aliases used: DB - database aliases; UL - usrloc aliases
- default: none , ALIASES_TYPE="DB"
control engine: RPCFIFO
 - default RPCFIFO
CTLENGINE="RPCFIFO"

## path to FIFO file for engine RPCFIFO
# RPCFIFOPATH="/var/run/kamailio/kamailio_rpc_fifo"

## check ACL names; default on (1); off (0)
# VERIFY_ACL=1

## ACL names-if VERIFY_ACL is set,only the ACL names from below list are accepted
# ACL_GROUPS="local ld int voicemail free-pstn"

## check if user exists (used by some commands such as acl);
## - default on (1); off (0)

# VERIFY_USER=1

## verbose - debug purposes - default '0'
# VERBOSE=1

## do (1) or don't (0) store plaintext passwords
## in the subscriber table - default '1'

# STORE_PLAINTEXT_PW=0

Kamailio START Options

PID file path – default is: /var/run/kamailio/kamailio.pid

# PID_FILE=/var/run/kamailio/kamailio.pid

Extra start options – default is: not set

# example: start Kamailio with 64MB share memory: STARTOPTIONS="-m 64"
# STARTOPTIONS=

Kamailio.cfg

config files are used to customize and deploy SIP services since each and every SIP packet is route based on policies specified in conf file ( routing blocks ). Location when installed from source – /usr/local/etc/kamailio/kamailio.cfg , when installed from package – /etc/kamailio/kamailio.cfg

The features in config file :-

  • User authentication

Kamailio doesn’t have user authentication by default , so to enable it one must

#!define WITH_MYSQL
#!define WITH_AUTH

kamdbctl tool is to be used for creating and managing the database.

kamdbctl create

Kamctl is used for adding subscriber information and password.

kamctl add altanai1 123
mysql: [Warning] Using a password on the command line interface can be insecure.
MySQL password for user 'kamailio@localhost': 
mysql: [Warning] Using a password on the command line interface can be insecure.
new user 'altanai1' added

More details in Tools section below .

  • IP authorization
  • accounting
  • registrar and location servicesTo have persisant location enabled so that records are not lost once kamailio are restarted , we need to save it to database and reload when restarting
#!define WITH_USRLOCDB
  • attacks detection and blocking (anti-flood protection)
  • NAT traversal

requires RTP proxy for RTP relay. NAT traversal support can be set by

#!define WITH_NAT
  • short dialing on server
  • multiple identities (aliases) for subscribers
  • multi-domain support
  • routing to a PSTN gateway
  • routing to a voicemail server
  • TLS encryption
  • instant messaging (pager mode with MESSAGE requests)
  • presence services

Kamailio (OpenSER) SIP Server v4.3- default configuration script

Several features can be enabled using ‘#!define WITH_FEATURE’ directives:

debugger params

define WITH_DEBUG
...
#!ifdef WITH_DEBUGs
modparam("debugger", "cfgtrace", 1)
#!endif

To run in debug mode: 

#!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

To enable mysql: 

define WITH_MYSQL
...
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif

To enable authentication

enable mysql
define WITH_AUTH

To enable IP authentication execute: enable mysql ,  enable authentication ,  define WITH_IPAUTH and  add IP addresses with group id ‘1’ to ‘address’ table.

To enable persistent user location :

enable mysql
define WITH_USRLOCDB

To enable presence server :

enable mysql
define WITH_PRESENCE

To enable nat traversal :

define WITH_NAT

RTPProxy

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

PSTN gateway

To enable PSTN gateway routing :

define WITH_PSTN

set the value of pstn.gw_ip

check route[PSTN] for regexp routing condition.

#!ifdef WITH_PSTN
pstn.gw_ip = "" desc "PSTN GW Address"
pstn.gw_port = "" desc "PSTN GW Port"
#!endif

Database aliases lookup

To enable database aliases lookup :

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
...
 #!ifdef WITH_MULTIDOMAIN# - the value for 'use_domain' parameter
 #!define MULTIDOMAIN 1
 #!else
 #!define MULTIDOMAIN 0
 #!endif

To enable TLS support execute:

adjust CFGDIR/tls.cfg as needed
define WITH_TLS

To enable XMLRPC support :

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
...
      
route[REQINIT] { 
  #!ifdef WITH_ANTIFLOOD 
  # flood detection from same IP and traffic ban      
  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 
.. 
}

To block 3XX redirect replies execute:

define WITH_BLOCK3XX

To enable VoiceMail routing :

define WITH_VOICEMAIL

set the value of voicemail.srv_ip and adjust the value of voicemail.srv_port

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

To enhance accounting execute:

enable mysql
define WITH_ACCDB

add following columns to database

define WITH_MYSQL
define WITH_AUTH
define WITH_USRLOCDB

#!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

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

Include Local Config If Exists

import_file “kamailio-local.cfg”

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 
 #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" 
 #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" 
 # !endif



 # 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 

 !substdef "!MY_IP_ADDR!!g"
 #!substdef "!MY_DOMAIN!!g" 
 #!substdef "!MY_WS_PORT!8080!g"
 #!substdef "!MY_WSS_PORT!4443!g"
 #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g"
 #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g"</pre>
 
 #!define WITH_WEBSOCKETS
disable TCP (default on)
#disable_tcp=yes
enable_sctp = 0

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”

//port to listen
port=5060
#!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

Modules Section

set paths to location of modules (to sources or installation folders)

#!ifdef WITH_SRCPATH
mpath="modules/"
#!else
mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/"
#!endif

Load modules

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
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

#!ifdef WITH_WEBSOCKETS
loadmodule "xhttp.so"
#loadmodule "websocket.so"
loadmodule "nathelper.so"
#!endif

modules params for mi_fifo,ctl, tm and rr

----- mi_fifo params -----
#modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")

----- ctl params -----
#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", 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)

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", "use_domain", MULTIDOMAIN)
#!endif

permissions params

#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!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)
modparam("domain", "register_myself", 1)
#!endif

presence params

#!ifdef WITH_PRESENCE
modparam("presence", "db_url", DBURL)
!endif

presence_xml params

modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)

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)

tls params

#!ifdef WITH_TLS
modparam("tls", "config", "/etc/kamailio/tls.cfg")
#!endif

pike params

#!ifdef WITH_ANTIFLOOD
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
#!endif

htable params

ip ban htable with autoexpire after 5 minutes

modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")

xmlrpc params

modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")

nathelper params

#!ifdef WITH_WEBSOCKETS
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
#!endif

Note: leaving NAT pings turned off here as nathelper is <em>only</em> being used for&nbsp;WebSocket connections. NAT pings are not needed as WebSockets have&nbsp;their own keep-alives.

Routing Logic

Main SIP request routing logic processing of any incoming SIP request starts with this route . Read more on Kamailio Call routing and Control

request_route { 
    # per request initial checks 
    route(REQINIT);

    #!ifdef WITH_WEBSOCKETS 
        if (nat_uac_test(64)) {     
            force_rport();     
            if (is_method("REGISTER")) {          
                fix_nated_register();     
            } else {         
                fix_nated_contact();         
                if (!add_contact_alias()) {  
                   xlog("L_ERR", "Error aliasing contact \n");             
                   sl_send_reply("400", "Bad Request");                    
                   exit;         
                }     
            } 
        } 
    #!endif

    # 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 
    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 
    route(LOCATION); 
}

Wrapper for relaying requests

enable additional event routes for forwarded requests – serial forking, RTP relaying handling, a.s.o.

route[RELAY] {    
    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 (!mf_process_maxfwd_header("10")) {     
   sl_send_reply("483","Too Many Hops");     
   exit; 
} 

if(is_method("OPTIONS") &amp;&amp; uri==myself &amp;&amp;; $rU==$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

sequential request withing a dialog should take the path determined by record-routing

route[WITHINDLG] {
    if (!has_totag()) return;
    if (has_totag()) {
        if (loose_route()) {
            #!ifdef WITH_WEBSOCKETS
            if ($du == "") {
                if (!handle_ruri_alias()) {
                    xlog("L_ERR", "Bad alias <$ru>\n");
                    sl_send_reply("400", "Bad Request");
                    exit;
                }
            }
            #!endif
         }
     exit;
     }

    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.
            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

tbd

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 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;
}

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 authentication

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 signaling 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 routing enabled but pstn.gw_ip not defined\n");
    return;
  }

  # route to PSTN dialed numbers starting with '+' or '00'(international format)
  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] {
if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents
    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($sel(cfg_get.voicemail.srv_ip))) {
    xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n");
    return;
}
if(is_method("INVITE")) {
    if($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
}

Supports pseudo-variables to access and manage parts of the SIP messages and attributes specific to users and server. Transformations to modify existing pseudo-variables, accessing only the wanted parts of the information.

Already has over 1000 parameters, variables and functions exported to config file. Supports runtime update framework – to avoid restarting the SIP server when needing to change the config parameters.

kamctl

Manage kamailio from command line, providing lots of operations, such as adding/removing/updating SIP users, controlling the ACL for users, managing the records for LCR or load balancing, viewing registered users and internal statistics, etc. When needed to interact with Kamailio, it does it via FIFO file created by mi_fifo module.

kamdbctl

Helps to configure and database needed by kamailio . First we need to select a database engine in the kamctlrc file by DBENGINE parameter .

Valid values are: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT.

The tool can be used to create and manage the database structure needed by Kamailio, therefore it should be immediately after Kamailio installation, in case you plan to run Kamailio with a database backend.

kamcmd

send RPC commands to Kamailio from command line, requires  ctl module

siremis

web management interface for Kamailio, written in PHP , AJAX , web 2.0 using MVC architecture

  • system and database administration tools for Kamailio SIP Server
  • subscriber, database aliases and speed dial management
  • location table view
  • dispatcher (load balancer), prefix-domain translation and least cost routing (lcr) management
  • access control lists (user groups) and permissions management
  • accounting records and missed calls vies
  • manage call data records (generated from acc records)
  • hash table, dial plan table and user preferences table management
  • offline message storage, presence service and sip trace views
  • communication with Kamailio SIP Server via XMLRPC ,  JSONRPC
  • communication with FreeSWITCH via event socket
  • create and display charts from statistic data stored by Kamailio
  • user location statistics charts
  • SIP traffic load charts
  • memory usage charts
  • accounting records charts and summary table
  • SQL-based CDR generation and rating billing engine

Covered in this article

  • Internal architecture
  • Configuration language
  • least cost routing
  • load balancing
  • traffic dispatching
  • DID routing
  • prefix based routing
  • SIP trunks and peering
  • traffic shaping
  • topology hiding
  • flood detection
  • scanning attacks prevention
  • anti-fraud policies
  • SQL and noSQL connectors
  • enum and DNS based routing
  • authentication and authorization
  • secure communication (TLS)
  • registration and location services
  • accounting and call data records
  • call control – redirect, forward, baring
  • redundancy and scalability
  • high availability and failover
  • websockets and webrtc

References :

  • [1] Henning Westerholt – Kamailio project-1&1 Internet AG ( 2009 )

Proxying Media Streams via Kamailio’s RTP Proxy


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