This document introduces Asterisk 11 and discusses its new features including improved support for WebSockets, WebRTC, and XMPP through Google/Jingle. It explains that Asterisk is an open source communications platform that has been in development since 1999 and is used widely for building voice over IP systems and applications. Asterisk 11 focuses on stability through architectural fixes and optimizations while adding some new capabilities like SIP over WebSockets to help enable real-time communications through browsers using emerging WebRTC standards.

1. Introducing Asterisk 11
WebSockets, WebRTC and More
Steve Sokol
Asterisk Marketing & Evangelism

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What is Asterisk?
§ Communications Engine
§ Open Source Project
§ 2+ Million Downloads Each Year
§ Over 1 Million Production Deployments
§ 85,000 Registered Community Members
§ Under Development Since 1999

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What Does Asterisk Do?
§ Plumbing:
§ Protocol Implementations
§ Media I/O
§ Media Management
§ Applications
§ Call Routing (“Dialplan”)
§ Voicemail
§ Conference Bridge
§ Queues
§ Automated Attendant
§ Directory
§ Call Parking
§ Session Management
§ Intermediary Services

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How Do You Use Asterisk?
§ As An Application Until Itself:
§ Hand-Crafted “One-Off” Implementations
§ Dialplan Script Programming + Configuration Files
§ PBX, VoIP Gateway, IVR, ACD, Etc.
§ As A Tool-Kit / Engine:
§ Use External Interfaces: AGI, AMI
§ “Wrap” Asterisk To Manage
§ Build Custom Applications & Solutions
§ Create Communications Products

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Elastix and Asterisk
§ Asterisk = Communications Engine
§ VoIP and PSTN Connectivity
§ Session (Call) Management
§ Basic Applications
§ Elastix = PBX / Business Logic
§ Call Flow and Treatment
§ End User Features
§ Intuitive Admin Interface
§ Additional UC Modalities (IM, Email, Calendar, Etc.)

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Asterisk Releases – 1.8 LTS
§ Asterisk 1.8 LTS
§ Released 2 Year Ago (AstriCon 2010)
§ Long Term Support (LTS)
§ Added:
§ Secure Calling (SRTP / SIP TLS)
§ IPv6 Support
§ Calendar Integration
§ XMPP Device State
§ Channel Event Logging
§ Google Voice / Chat / Talk Calling (Sort Of)
§ “ISDN” Features (AOC, CCSS, CPID)

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Asterisk Releases - Asterisk 10
§ Asterisk 10
§ Released ~1 year ago (AstriCon 2011)
§ Standard Release (1 Year Lifespan)
§ Added:
§ HD Media Engine
§ New Codecs
§ ConfBridge HD Audio / Video conferencing
§ T.38 Gateway
§ Text Message Routing

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Asterisk Releases – Certified Asterisk
§ Asterisk Release Policy 101
§ Mainline releases never get new features
§ Mainline releases contain everything
§ Mainline Asterisk Is Updated Frequently
§ About one new release each month
§ Good In Some Ways
§ Bugs get fixed quickly
§ Bad In Some Ways
§ Rapidly moving target
§ No new features

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Asterisk Releases – Certified Asterisk
§ Certified Asterisk
§ Based on the most recent LTS release
§ Removes unsupportable modules
§ Undergoes additional testing process
§ Includes limited number of high-value new features
§ Updated quarterly
§ Still Free Software: Same GPLv2 License

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Certified Asterisk
§ CERT1 Release
§ Based on 1.8 LTS
§ Back-ports SIP message routing from Asterisk 10
§ Adds additional APIs to Voicemail, Queues, Parking
§ Why?
§ Support For SLA Customers
§ Digium’s Asterisk Phones

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Phones For Asterisk
§ Digium D40, D50, D70 – Built For Asterisk Systems
§ Super Simple Provisioning
§ Discovery via mDNS / Bonjour
§ Uses SIP channel
§ Integrated Applications:
§ Visual Voicemail
§ Visual Call Parking
§ Queue Management
§ Presence / Status
§ Call Deflection / Direct VM

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Asterisk 11
§ LTS Release
§ 5 years of full support
§ 1 additional year of security support
§ Takes you to 2018
§ LTS = Stability NOT Features
§ Architectural-level bug fixes
§ Performance enhancements
§ Significant refactoring of existing features / functions

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New In Asterisk 11
§ Chan_motif – Google / Jingle / XMPP Done Right
§ Combines chan_google and chan_jingle into a single driver
§ Uses completely refactored XMPP engine (res_xmpp)
§ More stable / less difficult to keep up with Google
§ WebSockets Support For SIP
§ SIP uses multiple transports: UDP, TCP, TLS now WS
§ New protocol used by web applications for bi-directional,
asynchronous communications
§ Integrated into onboard HTTP server in Asterisk

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Why WebSockets?
§ WebRTC – A New Paradigm For Communications
§ Adds real-time communications to web browsers
§ Audio (Speakers / Microphone)
§ Video (Display / Camera)
§ Implements tools for media session management
§ NAT traversal (STUN, TURN, ICE)
§ Codecs (G.711, Opus)
§ Defines JavaScript APIs for media access, peer connection
§ Leaves the signaling protocol / process open to the application
developer

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SIP + WebSockets + WebRTC
§ Adding WebSockets to Asterisk enables
§ SIP over WebSockets (available in Asterisk 11)
§ XMPP / Jingle over WebSockets (future version)
§ ??? Over WebSockets (as soon as you like)
§ Other WebRTC changes
§ Improved RTP / SRTP handling
§ ICE support
§ SDP improvements

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This Is Revolutionary Stuff!
§ Instantly VoIP enabling every browser in the world
§ No software to install
§ Interoperability with existing VoIP technologies using RTP
§ SIP
§ Open standards: anyone can play
§ True unified communications
§ Voice + Video Calling
§ Screen Sharing
§ Conferencing

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Next Up: Asterisk 12
§ Asterisk is 13 years old
§ Asterisk has some architectural issues that need addressing
§ Asterisk has a very, very large installed base of users
§ How to make improvements without breaking things
§ Help us define the next release:
§ Join us at AstriDevCon: October 22, 2012 in Atlanta
§ One day prior to AstriCon

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Thanks!