Open Source Telephony Disruptive Solutions

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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.

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Open Source Telephony Disruptive Solutions

  1. 1. Title Open Source Telephony Disruptive Solutions IP Voice Meeting 2008 4-6 March 2008, Centro Cultural de Belém, Lisboa . Portugal Marco Mouta
  2. 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. 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. 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. 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. 6. Open Source Telephony Asterisk
  7. 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. 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. 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. 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. 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. 12. Open Source Telephony OpenSER & Asterisk
  13. 13. Open Source Telephony OpenSER & Asterisk
  14. 14. Open Source Telephony OpenSER & Asterisk Summary Interoperability Scalability Distributed systems Resilence Flexibility and self control
  15. 15. Innovative services ...new services are definitely changing the way we communicate!
  16. 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. 17. Innovative services Fax2Email & Email2Fax
  18. 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. 19. Innovative services Click to Talk facebook users can now be reached without ever providing their phone numbers!
  20. 20. Innovative services Click to Call
  21. 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. 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. 23. Thanks for your attention!
  24. 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. 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

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