This presentation is about: Open Source Telephony Solutions widely used across the world. Where do they live and grow across enterprises and public administrations. Disruptive and innovative services available for any one.

1. Title
Open Source Telephony Disruptive Solutions
IP Voice Meeting 2008
4-6 March 2008, Centro Cultural de Belém, Lisboa . Portugal
Marco Mouta

2. Introduction
This presentation is about:
Open Source Telephony Solutions widely used across the
world.
Where do they live and grow across enterprises and public
administrations.
Disruptive and innovative services available for any one.

3. Introduction
Outline:
Asterisk your VoIP telephony tool
Scalability and High level performance with OpenSER
Asterisk & Openser, The perfect marriage!
Innovative Services and Case Studies

4. Open Source Telephony
Asterisk
What is it?
Is the industry’s first Open Source telephony platform.
First release, Version 0.1.0, was created in December 1999
by Mark Spencer, Digium’s Asterisk Company founder.
Is the World’s leading Open Source PBX.
(quot;Asterisk passes million download milestonequot; - 19th December 2007)
Is a Great telephony toolkit for both VoIP and PSTN
environments.

5. Open Source Telephony
Asterisk
Extremely flexible and configurable, can act:
as a Media gateway.
Bridging the PSTN world into the world of IP telephony.
as a switch (PBX).
Pure IP Softswitch or hybrid PBX (PSTN and VoIP).
as an Application server.
VoiceMail to email, Conference Bridge, IVRs, Automated attendant and Telephony interface for your
Website!
in Call Centers.
Call Queues, Automatic Call Distribuition, Enables Remote IP Agents, Advanced skills-based routing,
predictive and bulk dialing.
in the network.
VoIP service providers, VoIP brokers, Carriers use it also for Voicemail systems, pre-paid calling solutions
as well as Call Shops and Cybercafes.

6. Open Source Telephony
Asterisk

7. Open Source Telephony
Asterisk
Extremely flexible and configurable, can act:
as a Media gateway.
Bridging the PSTN world into the world of IP telephony.
as a switch (PBX).
Pure IP Softswitch or hybrid PBX (PSTN and VoIP).
as an Application server.
VoiceMail to email, Conference Bridge, IVRs, Automated attendant and Telephony interface for your
Website!
in Call Centers.
Call Queues, Automatic Call Distribuition, Enables Remote IP Agents, Advanced skills-based routing,
predictive and bulk dialing.
in the network.
VoIP service providers, VoIP brokers, Carriers use it also for Voicemail systems, pre-paid calling solutions
as well as Call Shops and Cybercafes.

8. Open Source Telephony
Asterisk
Full Featured VoIP Solution
Extensions and DID’s for
H.323
users
Session Initiation Protocol
Follow-me, Huntgroups
(SIP)
VoiceMail,Conference
Media Gateway Control
Bridge,IVRs
Protocol (MGCP)
Call Queues, Agents and
Skinny Client Control
ACD
Protocol (SCCP)
Video Call suport as well
Inter-Asterisk eXchange
as Video Voicemail
(IAX)
E1/T1/BRI/Analogue
Jingle (Google Talk)
Telephony Interfaces

9. Open Source Telephony
OpenSER
What is it?
It’s a mature, flexible and scalable Open Source SIP server
quot;Cisco is using OpenSER in Cisco Service Nodes for Linksys One!quot;
Origin of OpenSER is the SIP Express Router (SER)
First release in Autumn 2002, FhG FOKUS research institute in Berlin, Germany.
In June 2005, two SER core developers and one main
contributor started OpenSER project.
(Bogdan-Andrei Iancu, Daniel-Constantin Mierla and Elena-Ramona Modroiu)
First release happened on the 14th June 2005, versioned
0.9.4
Source code forked from SER branch 0.9.0.

10. Open Source Telephony
OpenSER
Extremely flexible and configurable, can:
act as a SIP registrar, proxy server.
location server, redirect server.
act as a gateway to SMS and XMPP.
integrate with Jabber and provide you Instant Messaging
and Presence.

11. Open Source Telephony
OpenSER
Summary
Registrar, Proxy, Location and Redirect
Load balancing with failover
Geographical redundancy and distributed systems
Least cost routing
handle up to 5000 call setups per second!
(Load balancer-stateless mode)
handle up to 300 000 online subscribers!
(On systems with 4GB memory)
Do not forget this name, OpenSER, like SIP, is here to stay!

12. Open Source Telephony
OpenSER & Asterisk

13. Open Source Telephony
OpenSER & Asterisk

14. Open Source Telephony
OpenSER & Asterisk
Summary
Interoperability
Scalability
Distributed systems
Resilence
Flexibility and self control

15. Innovative services
...new services are definitely changing the way we
communicate!

16. Innovative services
Fax2Email & Email2Fax
What is this about?
Sending and receiving Faxes like we do with email!
Fax image, tipically is received or sent as an attachment
(pdf or tif file).
Instead of a single fax number for all company, received
faxes can be routed by DID.
I chose Hylafax & Asterisk because:
widely used across the world.
scalable and very customizable.
I have already deployed it.
So, why not present it to you?

17. Innovative services
Fax2Email & Email2Fax

18. Innovative services
Click to Talk
User Clicks the button.
1
Request is sent from user’s pc to
2
remote Asterisk server.
Asterisk answers the request and
3
dials a call to Facebook’s phone
contact previously configured.
Once the call is answered by
4
Facebook’s phone user both call
legs are bridge, and voilá!

19. Innovative services
Click to Talk
facebook users can now be reached without ever providing
their phone numbers!

20. Innovative services
Click to Call

21. Case study
Pop Idols - Portugal, 2004
Industry: Tele-Voting
Challenge:
Tele-voting system for multiple TV shows.
Permit near-real-time web acess to Calls data by television
stations personnel.
Solution:
One of the largest Asterisk installations.
Written to support the large call volumes typical of media
call-in.
Benefits:
1800 simultaneous calls.
Peak call volumes of 600,000 new call per hour.
A new level of price/performance.
Company: EVT - Emerging Voice Technology, Inc.

22. Case study
Universidad de Granada - Spain
Industry: Public Institution
Challenge:
Advanced VoIP telephony system to all University
extensions.
Text to speech (TTS) and speech recognition (ASR)
systems to easily implement new IVR features.
Call queueing and call recording.
Coexistence with legacy PBX
Solution:
2 Asterisk servers fully redundant with ISDN failover
Verbio TTS & ASR solutions fully compliant with Asterisk.
Benefits:
Up to 1000 extensions planned (350 now and growing) with
4 E1 lines.
Hi-Tech open source & cost-effective solution.
Avanzada7: the Asterisk official training and HW supplier
for Spain and Portugal.

23. Thanks for your attention!

24. Marco Mouta
Electronics and Computer Science Engineer, dCAP.
VoIP specialist engineer @ National Foundation for Scientific Computing
(Portuguese NREN)
email: marco.mouta@gmail.com

25. Bibliography
Bibliography
http://www.asterisk.org
http://www.openser.org
http://www.hylafax.org
http://jabberd.jabberstudio.org/2/
Jared Smith, Jim Van Meggelen, Leif Madsen, Asterisk: The Future of Telephony 2nd Edition, O’REILLY
Picture: quot;We are in this togetherquot;, from Portuguese artist Rodrigo Oliveira