presentacion kamailio uvigo_09262011

Laboratorio de Conmutación 12/13: Kamailio

By Antón Román 09/27/2011

Presentation

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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.

http://www.quobis.com/

Motivation

Quobis Networks SLUTodos los derechos reservados 3

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.


1 st Part

Biref introduction to telephony

Basics of IP Telephony (ToIP)


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



● Fix line between two endpoints

● Manual conmutation

● Automatic conmutation (relay-based)

● Computer-controlled automatic conmutation

● Digital conmutation only electronic

● IP Conmutation!!

1870

2010


Circuit conmutation paradigm (ends of 70s)


Digital conmutation (ISDN) (begining of 90s)


Packet conmutation paradigm (IP) (nowadays)



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.



What about mobile telephony?:

● Maybe the most important milestone in history of telephony.

● It was born as a circuit-oriented technology (GSM)

● It will be an all-IP network (4G→LTE) in a few years.

● ToIP concepts can be applied to Mobile telephony → convergence!

1970

2010


Basics of IP telephony


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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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

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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!

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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).

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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...

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2nd Part

SIP

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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...).

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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... ).

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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.

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4. SIP

SIP functions (cont.):

4. Call establishment (multimedia sessions).

5. Call management: call transfers, param renegotiation...

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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.

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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:[email protected] → sip:1234@<ip-server>sips:[email protected]

Domain is resolved by DNS. Three types of DNS records involved: SRV (protocol and port), NAPTR (protocols available) and A.

Copyright Quobis Networks 2010 – All rights reserved

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)


8. SIP

SIP responses:

Provisional (1xx): 100 Trying

180 Ringing

Successful (2xx): 200 OK

202 OK

Redirection (3xx): 302 Moved Temporaly

Request failure (4xx): 401 Unauthorized

404 Not found 407 Proxy Authentication Required

408 Request Timeout


SIP trapezoid

9. SIP


10. SIP

A Kamailio B

(Registrar local services)

……...... ……......

Register

200 Ok

Invite

100 Trying

Invite

100 Trying

180 Ringing 180 Ringing

200 Ok200 Ok

Ack

Media Session

Bye

200 Ok

Register + Call


11. SIP

REGISTER sip:10.1.20.245 SIP/2.0Via: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-83dafcc6From: "1234" <sip:[email protected]>;tag=a75abbe5b6e4cc6o1To: "1234" <sip:[email protected]>Call-ID: [email protected]: 33037 REGISTERMax-Forwards: 70Authorization: Digest username="1234",realm="10.1.20.245",nonce="4a389a3a000000984df34ebaabed674b40c0b27d8b354c1d",uri="sip:10.1.20.245",algorithm=MD5,response="216509e17700c67fb1b346675e0f46b2"Contact: "1234" <sip:[email protected]:5060>;expires=60User-Agent: Linksys/SPA941-5.1.5Content-Length: 0Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFERSupported: replaces

Initial REGISTER: endpoint->Kamailio


12. SIP

SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-83dafcc6From: "1234" <sip:[email protected]>;tag=a75abbe5b6e4cc6o1To: "1234" <sip:[email protected]>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.c168Call-ID: [email protected]: 33037 REGISTERWWW-Authenticate: Digest realm="10.1.20.245", nonce="4a389a750000009c100b4538aaabc0db79e49a68db0e7dba", stale=trueServer: Kamailio (1.5.0-notls (i386/linux))Content-Length: 0

401: Kamailio->endpoint


13. SIP

REGISTER sip:10.1.20.245 SIP/2.0Via: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-67a1ce3From: "1234" <sip:[email protected]>;tag=a75abbe5b6e4cc6o1To: "1234" <sip:[email protected]>Call-ID: [email protected]: 33038 REGISTERMax-Forwards: 70Authorization: Digest username="1234",realm="10.1.20.245",nonce="4a389a750000009c100b4538aaabc0db79e49a68db0e7dba",uri="sip:10.1.20.245",algorithm=MD5,response="34d004dc426d7b56d9d742dbc69aeb4c"Contact: "1234" <sip:[email protected]:5060>;expires=60User-Agent: Linksys/SPA941-5.1.5Content-Length: 0Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFERSupported: replaces

Authentiqued REGISTER: endpoint->Kamailio


14. SIP

SIP/2.0 200 OKVia: SIP/2.0/UDP 10.1.3.15:5060;branch=z9hG4bK-67a1ce3From: "1234" <sip:[email protected]>;tag=a75abbe5b6e4cc6o1To: "1234" <sip:[email protected]>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.4319Call-ID: [email protected]: 33038 REGISTERContact: <sip:[email protected]:5060>;expires=60Server: Kamailio (1.5.0-notls (i386/linux))Content-Length: 0

200: Kamailio->endpoint

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3rd Part

Kamailio

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Open Source ToIP projects.

✔ Clients: Ekiga, Twinkle, Jitsi.

✔ IPPBX: Asterisk (and derived projects), Freeswitch.

✔ Softswitch: KamailioSIP Router, OpenSIPS.

✔ MediaServer: SEMS, Asterisk, FreeSwitch.

✔ Gateway: (Asterisk || FreeSwitch) + hardware card

SIP Open Source projects.


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.

1. 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!

2. 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

3. 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).

4. Kamailio


Image got from www.kamailio.orgNote: in 3.x versions Database and MI API are not in the kernel.

5. Kamailio


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...

6. Kamailio. Configuration (1)


Configuration file has several parts:

1. General parameter setup : port, protocol...

2. Module load and module param setup:

loadmodule “permissions.so” ...modparam(“permissions”, “db_mode”, 1)

loadmodule “permissions.so” ...modparam(“permissions”, “db_mode”, 1)

3. Main route: every request that reaches Kamailio is executed in the main route.


route{...route[TO_KAMAILIO2]...

}

route{...route[TO_KAMAILIO2]...

}


4. Secondary routes (route[x]()): accesed from the main route (route()).

route[TO_KAMAILIO2]{ ...

t_on_failure(“1”)…

t_on_reply(“3”)}

route[TO_KAMAILIO2]{ ...

t_on_failure(“1”)…

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]{...

}



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]{ ...}


Fuente: Asipto

11. kamailio. Configuration (5)

Image got from www.kamailio.org


Fuente: Asipto


Image got from www.kamailio.org


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

QUOBIS NETWORKS

Pol. A Granxa P.26036400 Porriño (Spain)Tlf. +34 902 999 465Sip://sip.quobis.com

www.quobis.com

Creative Commons Attribution-NonCommercial 3.0 Unported License

http://www.quobis.com/

http://creativecommons.org/licenses/by-nc/3.0/

presentacion kamailio uvigo_09262011

Technology

ip networks

toip telephony

history of telephony

electronic ip conmutation

ip sip native ip protocol

protocols sip

packet conmutation paradigm

mobile telephony convergence