INGATE

From The Sip Trunking Experts

TMCNet:  Digium Announces Release of Asterisk 11 Open Source Communications Software

[October 31, 2012]

Digium Announces Release of Asterisk 11 Open Source Communications Software

HUNTSVILLE, Ala., Oct. 31, 2012 /PRNewswire via COMTEX/ -- Digium, Inc., the Asterisk Company, unveiled Asterisk 11 at its annual AstriCon users' conference meeting, a new release that features multiple contributions from the Asterisk developer community. Asterisk 11 includes a number of new features, including support for WebRTC over SIP and native integration with Digium's line of VoIP telephones. It is also a new Long Term Support (LTS) version of Asterisk, the world's most widely adopted open source communications engine.


As a Long Term Support release, Asterisk 11 is primarily focused on stability, performance and security, with a relatively short list of new features. LTS releases receive four years of support, with an additional year of security maintenance. Under this release plan, Asterisk 11 will be supported through 2016.

Significant new features include: WebSockets SIP Transport - WebRTC/RTCWEB brings real-time communications to web browsers. The new WebSockets transport for the Asterisk SIP channel allows browser-based SIP clients to connect with Asterisk and establish media sessions.

DTLS-SRTP Support - A secure transport for RTP media streams used by WebRTC and SIP endpoints.

ICE, STUN and TURN Support - A set of related technologies for establishing live media streams between software agents running behind network address translators (NATs) and firewalls. ICE, STUN and TURN have been incorporated into the Asterisk RTP engine as part of the effort to support WebRTC.

Motif - A new channel driver for supporting the Jingle protocol and Google Talk. Motif combines functions previously spread across multiple channels, and makes use of a new and more standards-compliant XMPP implementation.

Asterisk 11 is currently available from the Asterisk.org web site. The Asterisk development community has already begun working on Asterisk 12. For more information and documentation, visit https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation.

To download the new release, visit: www.asterisk.org/downloads.

About Digium Digium®, Inc., provides Asterisk custom communications and Switchvox Unified Communications (UC) business phone systems that deliver enterprise-class features at a price businesses can afford. We are the creator, primary developer and sponsor of the Asterisk project, the world's most widely used open source communications software that turns an ordinary computer into a feature-rich voice communications server. With a community of more than 80,000 members worldwide, Asterisk has been used to create VoIP communication solutions in more than 170 countries. Since 1999, Digium has become the open source alternative to proprietary communication providers, giving people an innovative solution for business communications that increase productivity. Digium's wide range of business communications products is sold through a worldwide network of reseller partners. More information is available at: www.digium.com or www.asterisk.org.

The Digium logo, Digium, Asterisk, Asterisk SCF, Switchvox, AsteriskNOW, Asterisk Appliance and the Asterisk logo are trademarks of Digium, Inc. All other trademarks are property of their respective owners.

Media Contacts: Julie Webb Laura Kempke Digium Schwartz MSL Boston +1 (256) 428-6203 +1 (781) 684-0770 jwebb@digium.com digium@schwartzmsl.com SOURCE Digium, Inc.

[ Back To SIP Trunking Home's Homepage ]

Loading
Subscribe here for your FREE
SIP TRUNKING enewslettter.

Featured Partner

Featured Whitepapers

SIP Security for the Enterprise
Voice over IP (VoIP) is incorporated into a variety of computer networks, both public and private, and used for everyday transactions and communications among carriers, businesses, government agencies...

Voice-Optimized Network Delivers Premier Call Experience
Customers equate call quality with business quality. Real-time communication, interpersonal interaction, and the cordial tone of a call center representative can create a positive impression of your business that no email can match.

Featured Datasheets

Ingate SIParator E-SBCs
Adopting SIP is a simple process with the Ingate SIParator, the secure enterprise session border controller (E-SBC). The SIParator makes secure SIP communications - including VoIP,SIP trunking and more - possible while working seamlessly with your existing network firewall.

Ingate Firewalls
Everyone is talking about enterprise usage of VoIP, instant messaging and other types of realtime communications including presence and conferencing.

SIP Trunk Solutions for Service Providers
The award-winning Ingate Firewall and Ingate SIParator deliver a high quality, reliable SIP trunk connection between the customer's IP-PBX and the service provider network, and solve interoperability issues to simplify deployments and support for remote diagnosis of reported issues.

Featured Case Studies

Case Study - Haiti
With this solution our doctors were able to reach anywhere in the world quickly and easily, to get consults from colleagues, facilitate treatment, order supplies 'on the fly' and also help victims report back to families. The solution from Business Mobility Systems and Ingate worked immediately.

Case Study - Turkish Petroleum
"With one Ingate at the customer headquarters, all of their remote workers can enjoy the benefits of Turkish Petroleum's IP-PBX from anywhere in the world.

Case Study - GCM
"The Ingate solution overcomes the NAT traversal hurdle and makes it possible to connect customers to our VoIP service, without the need of costly leased lines or support intensive VPN tunnels.

Case Study - Kool Smiles
"Ingate's solutions provide the advanced level of security necessary for this medical environment and the tools to make the interface to the service provider hassle-free."

Case Study - NMSAS
"This solution really saved the day for us," said Paul Mercier, IT Administrator, NMSAS. "We needed to leverage SIP trunks to reduce costs, and they had to work with our existing Microsoft OCS investment."

Discover Leisure Connects Remote Users to its IP-PBX
Discover Leisure is one of the largest resellers of caravans and motor homes in the UK. With 15 branch of?

Featured eBOOKS

Internet+: The Way Toward Global Unified Communication
Connecting the telephony of the enterprise PBX or Unified Communications (UC) system using SIP trunks instead of conventional telephone lines has been very successful in recent years.

What is SIP Trunking? Edition 2
SIP trunking is becoming more of a focus for service providers. One key issue many service providers face when deploying SIP trunks is NAT, or Network Address Translation, traversal.

What is SIP Trunking? Edition 1
A vast resource for information about all things SIP - including SIP, security, VoIP, SIP trunking and Unified Communications.

Featured Videos

E-SBCs AS The Demarcation Point:
Ingate's Steve Johnson talks to Erik Linask about the role session border controller plays as the demarcation point at...

Demystifying DPI
How can deep packet inspection protect your SIP traffic as well as your entire network?

Featured Resources

What's New

Presenting the New Ingate/Intertex Website:
Internet+ is an extended Internet access allowing high quality SIP (Session Initiation Protocol) based real-time person-to-person communication, everywhere and for any application. It applies to both fixed and mobile networks ...

Featured Blogs

Featured Webinars

Secure SIP Trunking:
What You Need to Know

Successfully Deploying Enterprise SIP Trunking:
Tools and Techniques for Overcoming Common Roadblocks

Featured Podcasts

Getting the Most Out of Your SIP Trunks:
Ingate's Steve Johnson and TMC's Erik Linask discuss how best practices forgetting the most out of SIP Trunking services and common pitfalls to avoid.