Quobis presented information on IP telephony and SIP. They discussed the history and evolution of telephony from fixed lines to mobile networks. IP telephony offers advantages like reusing existing infrastructure and open source projects. SIP is the dominant signaling protocol and allows for establishing multimedia sessions through signaling and negotiation of session parameters. Kamailio is an open source SIP softswitch that can perform proxy and registrar roles and is high performance software used in commercial VoIP networks.

1. Laboratorio de Conmutación
12/13: Kamailio
By Antón Román 09/27/2011

2. Presentation
Quobis, who we are: www.quobis.com
● Quobis is a Engineering company focused on VoIP technologies.
● Founded in 2006 by Teleco Uvigo alumnus.
● Premises at O Porriño.
● 90% engineers.
● Main customers: operators, call centers and enterprises
● Partnerships: AcmePacket, Aheeva, SIPWise and Iptego.
● Own products: MTRP, SecVoID, MCU,...
● R&D activities. Currently involved in Avanza and FP7 projects.
2

3. Motivation
Why are we doing this?
●
We strongly believe University must be linked to real world.
● Part of our contribution to Kamailio Open Source Project.
● Professionals with qualified VoIP skills are demanded.
● Close cooperation with university in R&D projects.
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4. st
1 Part
Biref introduction to telephony
Basics of IP Telephony (ToIP)
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5. Introduction to telephony.
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6. Introduction to telephony.
● 1833 Samuel Morse invents telegraph
● 1871 Antonio Meuci invents telephone
● 1876 Alexander Graham Bell patents telephone
● 1877 First telephone call in Spain
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7. Introduction to telephony.
1870
● Fix line between two endpoints
● Manual conmutation
● Automatic conmutation (relay-based)
● Computer-controlled automatic conmutation
● Digital conmutation only electronic
● IP Conmutation!!
2010
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8. Introduction to telephony.
Circuit conmutation paradigm (ends of 70s)

9. Introduction to telephony.
Digital conmutation (ISDN) (begining of 90s)

10. Introduction to telephony.
Packet conmutation paradigm (IP) (nowadays)

11. Introduction to telephony.
What advantages does ToIP offer?:
●TDM and ISDN conmutation technologies
were a monopoly of big corporations:
operators and makers.
● It allows to re-use knowhow and
infrastructure → high speed evolution
●It allows to re-use generic hardware →
reduces coste.
● Open Source ToIP projects are born.
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12. Introduction to telephony.
What about mobile telephony?:
●Maybe the most important milestone in
history of telephony. 1970
●It was born as a circuit-oriented technology
(GSM)
●It will be an all-IP network (4G→LTE) in a few
years.
2010
●ToIP concepts can be applied to Mobile
telephony → convergence!
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13. Basics of IP telephony
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14. 1. Basics of IP telephony
VoIP (Voice over IP) technologies
allow multimedia data transmission
(voice, video, IM... ) over IP networks.
ToIP (Telephony over IP) means
telephony systems implemented with VoIP
technology.
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15. 2. Basics of IP telephony
VoIP = Signaling Protocols (SIP) +
Voice transport protocols (RTP/RTCP)
There are more signaling protocols
besides SIP: H323, Skype(*), MGCP, IAX,
Skinny... however SIP is and will be the
most used in short and middle term.
(*) closed protocol
15

16. 4. Basics of IP telephony
Normally signaling and voice are sent separately
(different protocols and ports).
This gives flexibility to protocols and allows them
to be adapted to new codecs and future multimedia
requirements:
● H323 → TDM signaling over IP
● SIP → native IP protocol → easily readable!
16

17. 5. Basics of IP telephony
Multimedia sessions(1)
Audio and video: RTP/UDP + RTCP (for QoS, optional)
Codecs: G711a (64Kbps), G711u (64Kbps), G729 (8 or 13 Kbps, licensed),
GSM (13 Kbps)...
Multimedia session params are agreed during SIP call establishment
through SDP (Session Description Protocol).
SDP message are attached to SIP messages (2 protocols, 1 text
message).
17

18. 5. Basics of IP telephony
Multimedia sessions (2)
RTP (Real Time Protocol) designed to transmit real time
data→ not only voice or video
It can be used to transmit desktop sessions, medical
monitoring data, share prices...
18

19. nd
2 Part
SIP
19

20. 1. SIP
SIP (Session Initiation Protocol)
Protocol defined in RFC3261.
RFC(Request For Comments) are published by
IETF (Internet Engineering Task Force) and
define Internet standards. Many actors are
involved (academia, makers, operators...).
20

21. 2. SIP
Since it is based on RFC3261, many others
RFCs add new functionality: presence, IM...
SIP uses clear text messages (like HTTP,
actually it's based on it).
Designed to establish any type of
multimedia session (voice, videocall,
videostreaming, electrocardiogram values... ).
21

22. 3. SIP
SIP functions:
1.­ User location: users register themselves to say the IP/port pair
where they are listening to SIP traffic.
2.­ Multimedia parameter negotiation: it allows to negotiate session
parameters (codecs, packetization time, ports...).
3.­ User availability: it allows to publish and notify info about user
availability, and subscribe to user status.
22

23. 4. SIP
SIP functions (cont.):
4.­ Call establishment (multimedia sessions).
5.­ Call management: call transfers, param re­negotiation...
23

24. 5. SIP
SIP Roles
User Agent ­ examples: a SIP phone or a Media Server. It can play two
roles:
User Agent Client (UAC) – User Agent which sends a request
User Agent Server (UAS) ­ User Agent which answers a request
Redirect Server ­ User Agent Server which redirect requests
Proxy – sends request on behalf of users. It is what users 'see' of a SIP
network .
Registrar ­ accepts REGISTER messages and stores user location in a
Database.
24

25. 6. SIP
SIP transport
SIP messages can be sent over TCP (port 5060) and UDP
(port 5060). Normally UDP is used.
It can also be transported over TLS.
SIP URI: identifies and locates a user.
sip:user@domain.com → sip:1234@<ip-server>
sips:user@domain.com
Domain is resolved by DNS. Three types of DNS records
involved: SRV (protocol and port), NAPTR (protocols
available) and A.
25

26. 7. SIP
SIP messages
Requests:
● REGISTER: register users
● INVITE: session establishment
● BYE: end of session
● ACK: acknowledge (INVITE, 407,...)
● SUBSCRIBE: event subscription
● INFO: transmit info during a call (DTMF)
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27. 8. SIP
SIP responses:
Provisional (1xx): Redirection (3xx):
100 Trying 302 Moved Temporaly
180 Ringing
Request failure (4xx):
Successful (2xx): 401 Unauthorized
200 OK 404 Not found
202 OK 407 Proxy Authentication Required
408 Request Timeout
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28. 9. SIP
SIP trapezoid
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29. 10. SIP
Register + Call
A ……...... Kamailio ……...... B
(Registrar local services)
Register
200 Ok
Invite Invite
100 Trying 100 Trying
180 Ringing 180 Ringing
200 Ok 200 Ok
Ack
Media Session
Bye
200 Ok
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30. 11. SIP
Initial REGISTER: endpoint->Kamailio
REGISTER sip:10.1.20.245 SIP/2.0
Via: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-83dafcc6
From: "1234" <sip:1234@10.1.20.245>;tag=a75abbe5b6e4cc6o1
To: "1234" <sip:1234@10.1.20.245>
Call-ID: 14221856-8faf084d@10.1.3.15
CSeq: 33037 REGISTER
Max-Forwards: 70
Authorization: Digest
username="1234",realm="10.1.20.245",nonce="4a389a3a00000098
4df34ebaabed674b40c0b27d8b354c1d",uri="sip:10.1.20.245",algori
thm=MD5,response="216509e17700c67fb1b346675e0f46b2"
Contact: "1234" <sip:1234@10.1.3.15:5060>;expires=60
User-Agent: Linksys/SPA941-5.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER
Supported: replaces
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31. 12. SIP
401: Kamailio->endpoint
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-83dafcc6
From: "1234" <sip:1234@10.1.20.245>;tag=a75abbe5b6e4cc6o1
To: "1234"
<sip:1234@10.1.20.245>;tag=f8f2ab2c1295e90ed7dbb499b30f44b
2.c168
Call-ID: 14221856-8faf084d@10.1.3.15
CSeq: 33037 REGISTER
WWW-Authenticate: Digest realm="10.1.20.245",
nonce="4a389a750000009c100b4538aaabc0db79e49a68db0e7db
a", stale=true
Server: Kamailio (1.5.0-notls (i386/linux))
Content-Length: 0
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32. 13. SIP
Authentiqued REGISTER: endpoint->Kamailio
REGISTER sip:10.1.20.245 SIP/2.0
Via: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-67a1ce3
From: "1234" <sip:1234@10.1.20.245>;tag=a75abbe5b6e4cc6o1
To: "1234" <sip:1234@10.1.20.245>
Call-ID: 14221856-8faf084d@10.1.3.15
CSeq: 33038 REGISTER
Max-Forwards: 70
Authorization: Digest
username="1234",realm="10.1.20.245",nonce="4a389a750000009c
100b4538aaabc0db79e49a68db0e7dba",uri="sip:10.1.20.245",algor
ithm=MD5,response="34d004dc426d7b56d9d742dbc69aeb4c"
Contact: "1234" <sip:1234@10.1.3.15:5060>;expires=60
User-Agent: Linksys/SPA941-5.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER
Supported: replaces
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33. 14. SIP
200: Kamailio->endpoint
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-67a1ce3
From: "1234" <sip:1234@10.1.20.245>;tag=a75abbe5b6e4cc6o1
To: "1234"
<sip:1234@10.1.20.245>;tag=f8f2ab2c1295e90ed7dbb499b30f44b
2.4319
Call-ID: 14221856-8faf084d@10.1.3.15
CSeq: 33038 REGISTER
Contact: <sip:1234@10.1.3.15:5060>;expires=60
Server: Kamailio (1.5.0-notls (i386/linux))
Content-Length: 0
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34. rd
3 Part
Kamailio
34

35. Open Source ToIP projects.
SIP Open Source projects.
Clients: Ekiga, Twinkle, Jitsi.
✔
IP­PBX: Asterisk (and derived projects), Freeswitch.
✔
Softswitch: Kamailio­SIP Router, OpenSIPS.
✔
MediaServer: SEMS, Asterisk, FreeSwitch.
✔
Gateway: (Asterisk || FreeSwitch) + hardware card
✔
35

36. 1. Kamailio
Kamailio (formerly known as OpenSER): is a SIP
softswitch. It can peform every SIP role (for us it will be
Proxy and Registar).
It is a high performance and robust software. It is used
by VoIP providers: Voztelecom (Spain) and 1&1
(Germany).
Kamailio is released under GPL license. Several
companies offer professional support.
Now it is part of the SIP Router project.
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37. 2. Kamailio
Kamailio can be installed on any Linux distribution.
Real test: 150 calls/s. Virtual
machine with 512MB RAM and
1,4GHz.
A dedicated server with 4GB RAM
could provide service to 300,000
users → Vigo!
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38. 3. Kamailio
Kamailio can be used in different scenarios:
● NAT Traversal: mediaproxy, rtpproxy, nathelper
● Presence server: presence, presence_xml
● Load balancer: dispatcher, path
● Instant Messaging: imc, xmpp
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39. 4. Kamailio
●Kamailio is based on modules with a kernel which
performs following tasks: memory management,
parsing and transport message.
● Kamailio's funcionality can be enriched with modules
(dynamic libraries, .so) included in the configuration.
● Configuration is just a file: /etc/kamailio/kamailio.cfg
●Modules param and users are provisioned in Database
(MySQL, Postgres, Oracle).
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40. 5. Kamailio
Image got from www.kamailio.org
Page 40
Note: in 3.x versions Database and MI API are not in the kernel.
Copyright Quobis Networks 2010 – All rights reserved

41. 6. Kamailio. Configuration (1)
Configuration
● Routing behaviour is configured in kamailio.cfg.
●Kamailio must be restarted to apply in changes
kamailio.cfg.
●Variable params are configured in Database and can
be reloaded in execution time: user provision, ACL,
LCR...
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42. 7. Kamailio. Configuration (2)
Configuration file has several parts:
1. General parameter setup : port, protocol...
2. Module load and module param setup:
loadmodule “permissions.so”
loadmodule “permissions.so”
...
...
modparam(“permissions”, “db_mode”, 1)
modparam(“permissions”, “db_mode”, 1)
3. Main route: every request that reaches Kamailio is
executed in the main route.
route{
route{
...
...
route[TO_KAMAILIO2]
route[TO_KAMAILIO2]
...
...
}}
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43. 8. Kamailio. Configuration (3)
4. Secondary routes (route[x]()): accesed from the
main route (route()).
route[TO_KAMAILIO2]{
route[TO_KAMAILIO2]{
...
...
t_on_failure(“1”)
t_on_failure(“1”)
……
t_on_reply(“3”)
t_on_reply(“3”)
}}
5. Error route route{}: executed when Kamailio
receives an error reply to a sent request.
failure_route[1]{
failure_route[1]{
...
...
}}
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44. 10. Kamailio. Configuration (4)
6. Reply routes (onreply_route[x]()): executed
when a 2xx reply to a sent request is received.
onreply_route[1]{
onreply_route[1]{
...
...
}}
7. Branch routes (branch_route[x]()): executed to
create a parallel route to send request to another server
branch_route[1]{
branch_route[1]{
...
...
}}
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45. 11. kamailio. Configuration (5)
Fuente: Asipto
Page 45
Image got from www.kamailio.org
Copyright Quobis Networks 2010 – All rights reserved

46. 12. Kamailio. Configuration (6)
Fuente: Asipto
Page 46
Image got from www.kamailio.org
Copyright Quobis Networks 2010 – All rights reserved

47. 13. Kamailio. Init.d and kamctl
How to launch Kamailio:
/etc/init.d/kamailio start/stop/restart/status
Kamctl: tool to control and monitor kamailio
● Check main stats: kamctl monitor
● Check uptime: kamctl fifo uptime
● Add user to Kamailio DB: kamctl add 101 secret101
Page 47
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48. QUOBIS
NETWORKS
Pol. A Granxa P.260
36400 Porriño (Spain)
Tlf. +34 902 999 465
Sip://sip.quobis.com
www.quobis.com
Creative Commons Attribution-NonCommercial 3.0 Unported License