- About VoIP
- What is VoIP and what it can do for you
- Introduction to VoIP (video)
- Why should you switch to VoIP services?
- Analog Telephony
- Digital Telephony
- What is SIP?
- How to start with VoIP telephony
- Web based VoIP
- How to choose a right VoIP provider?
- Wi-Fi network and VoIP
- VoIP Codecs
- Free sip account
- Confidential calls
- VPN: UDP or TCP?
- Mobile VoIP
- VoIP on your mobile
- Asterisk IP-PBX
- Who we are?
- How to start
- Free SIP account
- Configs
WebRTC from Google: making real-time communication free to implement
After Microsoft bought Skype for US$8.5 billion, Google has released a developer preview of WebRTC – an open framework enabling implementation of voice and video Real Time Communications in the browser with the use of HTML 5 and JavaScript APIs.
It’s possible that the Skype acquisition could mean restricting the client from technologies and devices other than Microsoft’s own and Apple’s FaceTime is also far from being free and open source. Microsoft’s and Apple’s intentions aside, efforts by Google – which is working with Mozilla and Opera to develop the technology – could bring big changes to the way we use voice and video in the web … no more external, memory consuming VoIP programs?
Google’s approach is highly understandable as their core investments are Chrome OS and upcoming Chromebooks. In general, the philosophy behind Chrome OS is to exclude the need for installing any non web-based external software, so WebRTC makes sense.
The use of WebRTC is based on BSD license (open source), because the key idea is to build a free, open and cross-platform standard for voice/video chats. It means WebRTC is supposed to work on Android and other mobile platforms too. Apparently, the code is based on Global IP Solutions (GIPS) technology, a company acquired by Google In 2010. GIPS worked on VoIP solutions for mobile devices, such as VideoEngine versions for Android, iOS and Windows Mobile.
Perhaps we shouldn’t expect to see browser-based Skype alternative any soon, but the Google Talk team has already been working on moving the software from iSAC to WebRTC.
In the meantime, it’s worth to take a look at how Ericsson tested RTC technology:
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More to read on VoIP
- About
- About Mark Spencer
- Asterisk SIP Media NAT
- Browser-based VoIP: web page code to call over IP (to your VoIP account)
- Choosing the right provider
- Cisco ATA186 notes
- Free sip account
- Grandstream Budgetone configuration manual
- How to start with VoIP telephony
- Multiplexing RTP Data and Control Packets on a Single Port
- On-line payments
- VoIP Codecs
- VPN: UDP or TCP?
- What is VoIP and what it can do for you
- Why should you switch to VoIP services?
Blogroll
- Asterisk™: The Definitive Guide (new window) “Asterisk has been emblematic of the way that open source software has changed business—and changed the world”
- Blog Jon FreeSWITCH VOIP SIP Asterisk Linux Open Source
- Business.com Business.com is one of the Web’s largest directories for business products and services
- Ubuntu how-to www.ubuntuka.com Miscellaneous Ubuntu Tips, Tricks and Hints
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