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kamailio:kamailio-mixed-ipv4-ipv6

Run your own SIP VoIP service on both IPv4 and IPv6

Main author:
    Daniel-Constantin Mierla <miconda [at] gmail.com>
      - founder Kamailio SIP Server project

At the date of publishing the article, the world celebrated IPv6 day, a good opportunity to show how to build your own SIP-based VoIP service to run on both IPv4 and IPv6 networks using open source Kamailio SIP server.

Kamailio has support for IPv6 since 2002, but since the presence of IPv6 networks was not a part of reality, its usage was quite limited in the past. Also we tested the core and the most used modules last year at the SIPit event in Sweden. Now the interest has increased substantially, a fact visible also in the subject of the topics discussed on project's forums.

The target with this tutorial is to have a service that is able to handle VoIP traffic:

  • in IPv4 networks
  • in IPv6 networks
  • bridge between IPv4 and IPv6 networks when needed

Kamailio itself is a SIP signaling server, therefore it will take care of bridging the SIP traffic. To bridge the media stream between IPv4 and IPv6, we will use RTPProxy, an application used so far mainly to help with NAT traversal in IPv4 networks.

Did I say NAT? Fortunately it will disappear soon, when (if) IPv6 will take the control in Internet communications.

About the environment

To make the guidelines for this tutorial, I built a local lab environment where I used:

  • Kamailio devel version running on Ubuntu 11.04 with next IP addresses:
    • IPv4: 192.168.178.26
    • IPv6: fec0:0:0:1001::1
  • Jitsi SIP softphone running on Mac OS X with next IP address:
    • IPv4: 192.168.178.21
  • Jitsi SIP softphone running on Ubuntu 10.10 with next IP address:
    • IPv6: fec0:0:0:1001::3

To install Kamailio devel version (to become 3.2.0 release in Autumn 2011), use following guidelines:

I used Kamailio development version since the plan is to include the IPv6 support in the default configuration file for the next major release 3.2.0, thus making my life easier in the near future.

However, the same concept can be applied to the latest stable v3.1.x, just use the appropriate functions from rtpproxy module - the development version introduced a new function rtpproxy_manage() which is a wrapper around existing functions in v3.1.x: rtpproxy_offer(), rtpproxy_answer() and unforce_rtp_proxy() – the parameters have the same meanings both versions.

Debian/Ubuntu users can install Kamailio development version from the APT repository (nightly builds), see:

Jitsi is a cross platform (Linux, Mac OS X and Windows) open source SIP softphone available at:

FYI: I tested also with Linphone on IPv6 and seemed to be fine – a side not: Linphone for Mac OS X didn't have support for video and I wanted to test also multi-streams IPv4-IPv6 bridging (audio and video in this case).

RTPProxy is available at:

  • for this article I installed RTPProxy from Ubuntu default APT repository with:
apt-get install rtpproxy

RTPProxy creates a special user (named rtpproxy) to run under unprivileged permissions. You have to allow Kamailio to communicate with it via control socket. One easy way is to edit /etc/init.d/rtpproxy and set the running user ID to kamailio since it is the only application accessing it, like:

USER=kamailio

Note that RTPProxy has to be started with special parameters in order o bridge between IPv4 and IPv6 addresses:

-l ADDR_IPV4 -6 /ADDR_IPV6

You have to change the /etc/init.d/rtpproxy to add these parameters.

Sample Scenario

I called using one Jitsi phone on IPv6 and one on IPv4, turning on video conversation – this is the most interesting case in my opinion: bridging real-time communication between IPv6 and IPv4 networks.

Here is the result - I is talking to me.

SIP Traffic

For the technical guys, I pasted below the relevant SIP signaling messages for this call (I stripped some of the SDP content since it was too big and irrelevant for this case – just codecs attributes).

Some remarks about the SIP traffic:

  • call comes in from IPv6 (frame [1] INVITE) with headers and SDP body filled with IPv6 addresses
  • when the call is forwarded by Kamailio to callee (frame [2]), you can observe:
    • presence of two Record-Route heders, one with Kamailio's IPv4 address and the other with Kamailio's IPv6 address
    • although incoming INVITE had SDP body with caller's IPv6 address, now the SDP body has RTPProxy IPv4 address since the INVITE is sent to IPv4 network
  • when call is answered, the 200ok SIP replies comes in from callee (frame [3]), with IPv4 addresses in SDP body
  • then the 200ok reply is relayed to caller (frame [4]), but now the SDP body has the IPv6 address of RTPProxy
  • the communication is ready to start, each side sending packets only in one type of network. SIP routing is driven by Record-Route headers and RTP by IP address in SDP body.
[1] fec0:0:0:1001::3.5060 > fec0:0:0:1001::1.5060

INVITE sip:102@[fec0:0:0:1001::1] SIP/2.0
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
To: <sip:102@[fec0:0:0:1001::1]>
Via: SIP/2.0/UDP [fec0:0:0:1001:0:0:0:3]:5060;branch=z9hG4bK-373838-0724cc1aa770fb3d7a70a456274bd649
Max-Forwards: 70
Contact: "103" <sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1]>
User-Agent: Jitsi1.0-beta1-nightly.build.3522Linux
Content-Type: application/sdp
Content-Length: 875

v=0
o=103 0 0 IN IP6 fec0:0:0:1001:0:0:0:3
s=-
c=IN IP6 fec0:0:0:1001:0:0:0:3
t=0 0
m=audio 5004 RTP/AVP 9 96 97 0 8 98 3 99 5 6 4 15 101
a=rtpmap:9 G722/8000
...


[2] 192.168.178.26.5060 > 192.168.178.21.5060

INVITE sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26 SIP/2.0
Record-Route: <sip:192.168.178.26;r2=on;lr=on;nat=v46>
Record-Route: <sip:[FEC0:0:0:1001:0:0:0:1];r2=on;lr=on;nat=v46>
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
To: <sip:102@[fec0:0:0:1001::1]>
Max-Forwards: 69
Contact: "103" <sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1]>
User-Agent: Jitsi1.0-beta1-nightly.build.3522Linux
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.178.26;branch=z9hG4bK0099.a33fcf56.0
Via: SIP/2.0/UDP [fec0:0:0:1001:0:0:0:3]:5060;rport=5060;branch=z9hG4bK-373838-da2adaf5a3c34a1de259916f79ba11b7
Content-Length: 881

v=0
o=103 0 0 IN IP4 192.168.178.26
s=-
c=IN IP4 192.168.178.26
t=0 0
m=audio 61536 RTP/AVP 9 96 97 0 8 98 3 99 5 6 4 15 101
a=rtpmap:9 G722/8000
...
a=nortpproxy:yes


[3] 192.168.178.21.5060 > 192.168.178.26.5060

SIP/2.0 200 OK
To: <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
Via: SIP/2.0/UDP 192.168.178.26;branch=z9hG4bK0099.a33fcf56.0,SIP/2.0/UDP [fec0:0:0:1001:0:0:0:3]:5060;rport=5060;branch=z9hG4bK-373838-da2adaf5a3c34a1de259916f79ba11b7
Record-Route: <sip:192.168.178.26;r2=on;lr=on;nat=v46>,<sip:[FEC0:0:0:1001:0:0:0:1];r2=on;lr=on;nat=v46>
CSeq: 2 INVITE
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
From: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
Contact: "102" <sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26>
User-Agent: Jitsi1.0-beta1-nightly.build.3521Mac OS X
Content-Type: application/sdp
Content-Length: 781

v=0
o=102 0 0 IN IP4 192.168.178.21
s=-
c=IN IP4 192.168.178.21
t=0 0
m=audio 5008 RTP/AVP 9 96 97 0 8 98 99 5 6 15 101
a=rtpmap:9 G722/8000
...


[4] fec0:0:0:1001::1.5060 > fec0:0:0:1001::3.5060

SIP/2.0 200 OK
To: <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
Via: SIP/2.0/UDP [fec0:0:0:1001:0:0:0:3]:5060;rport=5060;branch=z9hG4bK-373838-da2adaf5a3c34a1de259916f79ba11b7
Record-Route: <sip:192.168.178.26;r2=on;lr=on;nat=v46>,<sip:[FEC0:0:0:1001:0:0:0:1];r2=on;lr=on;nat=v46>
CSeq: 2 INVITE
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
From: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
Contact: "102" <sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26>
User-Agent: Jitsi1.0-beta1-nightly.build.3521Mac OS X
Content-Type: application/sdp
Content-Length: 805

v=0
o=102 0 0 IN IP6 fec0:0:0:1001::1
s=-
c=IN IP6 fec0:0:0:1001::1
t=0 0
m=audio 38516 RTP/AVP 9 96 97 0 8 98 99 5 6 15 101
a=rtpmap:9 G722/8000
...
a=nortpproxy:yes


[5] fec0:0:0:1001::3.5060 > fec0:0:0:1001::1.5060

ACK sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26 SIP/2.0
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/UDP [fec0:0:0:1001:0:0:0:3]:5060;branch=z9hG4bK-373838-cb66ec5365d63fe8e6f3f81ab0f66a7b
From: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
To: "102" <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
Max-Forwards: 70
Route: <sip:[FEC0:0:0:1001:0:0:0:1];r2=on;lr=on;nat=v46>,<sip:192.168.178.26;r2=on;lr=on;nat=v46>
Contact: "103" <sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1]>
User-Agent: Jitsi1.0-beta1-nightly.build.3522Linux
Content-Length: 0


[6] 192.168.178.26.5060 > 192.168.178.21.5060

ACK sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26 SIP/2.0
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.178.26;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP [fec0:0:0:1001:0:0:0:3]:5060;rport=5060;branch=z9hG4bK-373838-cb66ec5365d63fe8e6f3f81ab0f66a7b
From: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
To: "102" <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
Max-Forwards: 69
Contact: "103" <sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1]>
User-Agent: Jitsi1.0-beta1-nightly.build.3522Linux
Content-Length: 0



[7] 192.168.178.21.5060 > 192.168.178.26.5060

BYE sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1] SIP/2.0
CSeq: 1 BYE
From: <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
To: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
Max-Forwards: 70
Route: <sip:192.168.178.26;r2=on;lr=on;nat=v46>,<sip:[FEC0:0:0:1001:0:0:0:1];r2=on;lr=on;nat=v46>
Via: SIP/2.0/UDP 192.168.178.21:5060;branch=z9hG4bK-393630-97675b8dfcf6c8698398b150cdb24845
Contact: "102" <sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26>
User-Agent: Jitsi1.0-beta1-nightly.build.3521Mac OS X
Content-Length: 0



[8] fec0:0:0:1001::1.5060 > fec0:0:0:1001::3.5060

BYE sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1] SIP/2.0
CSeq: 1 BYE
From: <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
To: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
Max-Forwards: 69
Via: SIP/2.0/UDP [FEC0:0:0:1001:0:0:0:1];branch=z9hG4bK3099.f30b84.0
Via: SIP/2.0/UDP 192.168.178.21:5060;rport=5060;branch=z9hG4bK-393630-97675b8dfcf6c8698398b150cdb24845
Contact: "102" <sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26>
User-Agent: Jitsi1.0-beta1-nightly.build.3521Mac OS X
Content-Length: 0


[9] fec0:0:0:1001::3.5060 > fec0:0:0:1001::1.5060

SIP/2.0 200 OK
To: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
Via: SIP/2.0/UDP [FEC0:0:0:1001:0:0:0:1];branch=z9hG4bK3099.f30b84.0;received=fec0:0:0:1001:0:0:0:1,SIP/2.0/UDP 192.168.178.21:5060;rport=5060;branch=z9hG4bK-393630-97675b8dfcf6c8698398b150cdb24845
CSeq: 1 BYE
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
From: <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
Contact: "103" <sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1]>
User-Agent: Jitsi1.0-beta1-nightly.build.3522Linux
Content-Length: 0


[10] 192.168.178.26.5060 > 192.168.178.21.5060

SIP/2.0 200 OK
To: "103" <sip:103@[fec0:0:0:1001::1]>;tag=3891da55
Via: SIP/2.0/UDP 192.168.178.21:5060;rport=5060;branch=z9hG4bK-393630-97675b8dfcf6c8698398b150cdb24845
CSeq: 1 BYE
Call-ID: cc8f5a3f528f880e3caabbf11605b0c2@0:0:0:0:0:0:0:0
From: <sip:102@[fec0:0:0:1001::1]>;tag=7e28fdba
Contact: "103" <sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1]>
User-Agent: Jitsi1.0-beta1-nightly.build.3522Linux
Content-Length: 0

Location Records

The location records in Kamailio taken with kamctl looked like:

# kamctl ul show

Domain:: location table=512 records=2 max_slot=1
	AOR:: 102
		Contact:: sip:102@192.168.178.21:5060;transport=udp;registering_acc=192_168_178_26 Q=
			Expires:: 562
			Callid:: 54cf25d15fe668a1d542616093f99c70@0:0:0:0:0:0:0:0
			Cseq:: 2
			User-agent:: Jitsi1.0-beta1-nightly.build.3521Mac OS X
			Received:: sip:192.168.178.21:5060
			State:: CS_DIRTY
			Flags:: 0
			Cflag:: 2
			Socket:: udp:192.168.178.26:5060
			Methods:: 4294967295
	AOR:: 103
		Contact:: sip:103@[fec0:0:0:1001:0:0:0:3]:5060;transport=udp;registering_acc=[fec0:0:0:1001::1] Q=
			Expires:: 596
			Callid:: b5291cb60c79d65c65291364c7103138@0:0:0:0:0:0:0:0
			Cseq:: 2
			User-agent:: Jitsi1.0-beta1-nightly.build.3522Linux
			State:: CS_DIRTY
			Flags:: 0
			Cflag:: 64
			Socket:: udp:[FEC0:0:0:1001:0:0:0:1]:5060
			Methods:: 4294967295

Kamailio Configuration File

The configuration file for IPv4-IPv6 support is available for download further down. You have to copy it over /usr/local/etc/kamailio/kamailio.cfg.

Then you have to edit it and update the values for IPv4 and IPv6 addresses acording to your environment, search the lines:

#!define ADDR_IPV4 192.168.178.26
#!define ADDR_IPV6 [fec0:0:0:1001::1]

Then you are ready to go - start kamailio with this configuration file.

To check if it is listening on both IPv4 and IPv6, use kamctl tool:

# kamctl ps

Process::  ID=0 PID=4047 Type=attendant
Process::  ID=1 PID=4048 Type=udp receiver child=0 sock=192.168.178.26:5060
Process::  ID=2 PID=4049 Type=udp receiver child=1 sock=192.168.178.26:5060
Process::  ID=3 PID=4050 Type=udp receiver child=2 sock=192.168.178.26:5060
Process::  ID=4 PID=4051 Type=udp receiver child=3 sock=192.168.178.26:5060
Process::  ID=5 PID=4052 Type=udp receiver child=0 sock=[FEC0:0:0:1001:0:0:0:1]:5060
Process::  ID=6 PID=4053 Type=udp receiver child=1 sock=[FEC0:0:0:1001:0:0:0:1]:5060
Process::  ID=7 PID=4054 Type=udp receiver child=2 sock=[FEC0:0:0:1001:0:0:0:1]:5060
Process::  ID=8 PID=4055 Type=udp receiver child=3 sock=[FEC0:0:0:1001:0:0:0:1]:5060
Process::  ID=9 PID=4056 Type=slow timer
Process::  ID=10 PID=4057 Type=timer
Process::  ID=11 PID=4059 Type=MI FIFO
Process::  ID=12 PID=4063 Type=ctl handler
Process::  ID=13 PID=4064 Type=TIMER NH
Process::  ID=14 PID=4067 Type=tcp receiver child=0
Process::  ID=15 PID=4068 Type=tcp receiver child=1
Process::  ID=16 PID=4069 Type=tcp receiver child=2
Process::  ID=17 PID=4070 Type=tcp receiver child=3
Process::  ID=18 PID=4072 Type=tcp main process

You can spot easily the UDP receivers on IPv6 address.

kamailio.cfg

Here is the content you have to place in /usr/local/etc/kamailio/kamailio.cfg:

#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v3.2 - 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/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode: 
#     - define WITH_DEBUG
#
# *** 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
#
# *** 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 enable IPv6 routing execute:
#     - define WITH_IPV6
#     - define ADDR_IPV4 to server's IPv4 address
#     - define ADDR_IPV6 to server's IPv6 address
#     - enable nat traversal
#         * run RTPProxy in bridge mode between ADDR_IPV4 and ADDR_IPV6
#         * - RTPProxy options: -l ADDR_IPV4 -6 /ADDR_IPV6
#
# *** 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 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 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
 
####### Defined Values #########
 
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_NAT
#!define WITH_IPV6
#!define ADDR_IPV4 192.168.178.26
#!define ADDR_IPV6 [fec0:0:0:1001::1]
 
# *** 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.
#!define DBURL "mysql://openser:openserrw@localhost/openser"
#!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
 
#!define FLB_NATB 1
#!define FLB_NATSIPPING 2
#!define FLB_IPV6 6
#!define FLB_V4V6 7
 
####### Global Parameters #########
 
#!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"
 
#!ifdef WITH_IPV6
listen=ADDR_IPV4
listen=ADDR_IPV6
#!else
/* 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:10.0.0.10:5060
#!endif
 
/* port to listen to
 * - can be specified more than once if needed to listen on many ports */
port=5060
 
#!ifdef WITH_TLS
enable_tls=yes
#!endif
 
####### 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"
#!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
#!ifdef LOCAL_TEST_RUN
mpath="modules_k:modules"
#!else
mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"
#!endif
 
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
 
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.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
 
# ----------------- setting module-specific parameters ---------------
 
 
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_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 -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
 
 
# ----- 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)
 
 
# ----- 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;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=$fd;dst_ouser=$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", "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
 
 
# ----- speedial 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)
# use caching
modparam("domain", "db_mode", 1)
# 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 -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
 
 
#!ifdef WITH_NAT
# ----- rtpproxy params -----
# modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")
 
# ----- 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/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)
#!endif
 
####### Routing Logic ########
 
 
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {
 
	# per request initial checks
	route(REQINIT);
 
	# NAT detection
	route(NATDETECT);
 
	# handle requests within SIP dialogs
	route(WITHINDLG);
 
	### only initial requests (no To tag)
 
	# CANCEL processing
	if (is_method("CANCEL"))
	{
		if (t_check_trans())
			t_relay();
		exit;
	}
 
	t_check_trans();
 
	# 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);
 
	route(RELAY);
}
 
 
route[RELAY] {
 
	# enable additional event routes for forwarded requests
	# - serial forking, RTP relaying handling, a.s.o.
	if (is_method("INVITE|SUBSCRIBE")) {
		t_on_branch("MANAGE_BRANCH");
		t_on_reply("MANAGE_REPLY");
	}
	if (is_method("INVITE")) {
		t_on_failure("MANAGE_FAILURE");
	}
 
	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;
		}
	}
#!endif
 
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}
 
	if(!sanity_check("1511", "7"))
	{
		xlog("Malformed SIP message from $si:$sp\n");
		exit;
	}
}
 
# Handle requests within SIP dialogs
route[WITHINDLG] {
	if (has_totag()) {
		# sequential request withing a dialog should
		# take the path determined by record-routing
		if (loose_route()) {
			if (is_method("BYE")) {
				setflag(FLT_ACC); # do accounting ...
				setflag(FLT_ACCFAILED); # ... even if the transaction fails
			}
			if ( is_method("ACK") ) {
				# ACK is forwarded statelessy
				route(NATMANAGE);
			}
			route(RELAY);
		} else {
			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
					t_relay();
					exit;
				} else {
					# ACK without matching transaction ... ignore and discard
					exit;
				}
			}
			sl_send_reply("404","Not here");
		}
		exit;
	}
}
 
# Handle SIP registrations
route[REGISTRAR] {
	if (is_method("REGISTER"))
	{
		if(isflagset(FLT_NATS))
		{
			setbflag(FLB_NATB);
			# uncomment next line to do SIP NAT pinging 
			## setbflag(FLB_NATSIPPING);
		}
		if(af==INET6)
			setbflag(FLB_IPV6);
		if (!save("location"))
			sl_reply_error();
 
		exit;
	}
}
 
# USER location service
route[LOCATION] {
 
#!ifdef WITH_SPEEDIAL
	# search for short dialing - 2-digit extension
	if($rU=~"^[0-9][0-9]$")
		sd_lookup("speed_dial");
#!endif
 
#!ifdef WITH_ALIASDB
	# search in DB-based aliases
	alias_db_lookup("dbaliases");
#!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);
	}
}
 
# Presence server route
route[PRESENCE] {
	if(!is_method("PUBLISH|SUBSCRIBE"))
		return;
 
#!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;
}
 
# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
	if (is_method("REGISTER"))
	{
		# authenticate the REGISTER requests (uncomment to enable auth)
		if (!www_authorize("$td", "subscriber"))
		{
			www_challenge("$td", "0");
			exit;
		}
 
		if ($au!=$tU)
		{
			sl_send_reply("403","Forbidden auth ID");
			exit;
		}
	} else {
 
#!ifdef WITH_IPAUTH
		if(allow_source_address())
		{
			# source IP allowed
			return;
		}
#!endif
 
		# authenticate if from local subscriber
		if (from_uri==myself)
		{
			if (!proxy_authorize("$fd", "subscriber")) {
				proxy_challenge("$fd", "0");
				exit;
			}
			if (is_method("PUBLISH"))
			{
				if ($au!=$fU || $au!=$tU) {
					sl_send_reply("403","Forbidden auth ID");
					exit;
				}
				if ($au!=$rU) {
					sl_send_reply("403","Forbidden R-URI");
					exit;
				}
#!ifdef WITH_MULTIDOMAIN
				if ($fd!=$rd) {
					sl_send_reply("403","Forbidden R-URI domain");
					exit;
				}
#!endif
			} else {
				if ($au!=$fU) {
					sl_send_reply("403","Forbidden auth ID");
					exit;
				}
			}
 
			consume_credentials();
			# caller authenticated
		} else {
			# caller is not local subscriber, then check if it calls
			# a local destination, otherwise deny, not an open relay here
			if (!uri==myself)
			{
				sl_send_reply("403","Not relaying");
				exit;
			}
		}
	}
#!endif
	return;
}
 
# Caller NAT detection route
route[NATDETECT] {
	if(af==INET6)
		return;
 
#!ifdef WITH_NAT
	force_rport();
	if (nat_uac_test("19")) {
		if (is_method("REGISTER")) {
			fix_nated_register();
		} else {
			fix_nated_contact();
		}
		setflag(FLT_NATS);
	}
#!endif
	return;
}
 
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
	if (is_request()) {
		if(has_totag()) {
			if(check_route_param("nat=yes")) {
				setbflag(FLB_NATB);
			} else {
				if(check_route_param("nat=v46")) {
					setbflag(FLB_V4V6);
				}
			}
		}
	}
	if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)
			|| isbflagset(FLB_V4V6)))
		return;
 
	if(isbflagset(FLB_V4V6)) {
		if(af==INET6) {
			rtpproxy_manage("FAEI");
		} else {
			rtpproxy_manage("FAIE");
		}
	} else {
		if(af==INET6) {
			rtpproxy_manage("FAEE");
		} else {
			rtpproxy_manage("FAII");
		}
	}
 
	if (is_request()) {
		if (!has_totag()) {
			if(isbflagset(FLB_V4V6)) {
				add_rr_param(";nat=v46");
			} else {
				add_rr_param(";nat=yes");
			}
		}
	}
	if (is_reply()) {
		if(isbflagset(FLB_NATB)) {
			if(af==INET) {
				fix_nated_contact();
			}
		}
	}
#!endif
	return;
}
 
# Routing to foreign domains
route[SIPOUT] {
	if (!uri==myself)
	{
		append_hf("P-hint: outbound\r\n");
		route(RELAY);
	}
}
 
# 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;
	}
 
	$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
 
	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
 
# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
	if(!is_method("INVITE"))
		return;
 
	# check if VoiceMail server IP is defined
	if (strempty($sel(cfg_get.voicemail.srv_ip))) {
		xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
		return;
	}
	if($avp(oexten)==$null)
		return;
 
	$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
				+ $sel(cfg_get.voicemail.srv_port);
	route(RELAY);
	exit;
#!endif
 
	return;
}
 
# detect if IPv4 - IPv6 bridging is needed
route[IPV4V6] {
	if(!has_totag()) {
		if(af==INET6 && !isbflagset(FLB_IPV6)) {
			setbflag(FLB_V4V6);
		} else {
			if(af==INET && isbflagset(FLB_IPV6)) {
				setbflag(FLB_V4V6);
			}
		}
		return;
	}
}
 
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
	xdbg("new branch [$T_branch_idx] to $ru\n");
	route(IPV4V6);
	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")) {
		route(TOVOICEMAIL);
		exit;
	}
#!endif
}
If you installed Kamailio from APT repository, you have to adapt the mpath parameter to point to /usr/lib….

Remarks

With this kind of deployment, you can have SIP/VoIP conversations like:

  • IPv4 to IPv4
  • IPv6 to IPv6
  • IPv4 to IPv6
  • IPv6 to IPv4

The communication is not limited to voice and video, you can do Instant Messaging, Presence notifications of Desktop sharing as well.

The NAT concept is gone in IPv6, however the platform is intended to run in mixed environment, therefore you can still see NAT traversal handling logic, being just ignored in IPv6.



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