WebRTC (Web Real-Time Communication) is an HTML5 standard being drafted by the World Wide Web Consortium (W3C), with a mailing list created in April 2011.[1][2], and jointly in the IETF with a working group chartered in May 2011.[3] It is also the name of framework that was open sourced on June 1, 2011, which implements early versions of the standard and allows web browsers to conduct real-time communication.[4] The goal of WebRTC is to enable applications such as voice calling, video chat and P2P file sharing without plugins.
Category Archives: VOIP
Skype makes chats and user data more available to police
Skype, the online phone service long favored by political dissidents, criminals and others eager to communicate beyond the reach of governments, has expanded its cooperation with law enforcement authorities to make online chats and other user information available to police, said industry and government officials familiar with the changes.
via Skype makes chats and user data more available to police – The Washington Post.
Welcome to SIPp
SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also reads custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.
via Welcome to SIPp.
SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, … It is also very useful to emulate thousands of user agents calling your SIP system.
Jitsi (SIP Communicator)
Jitsi (previously SIP Communicator) is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.
Business Grade Instant Messaging and Presence
Microsoft® Lync® is an enterprise-ready unified communications platform. With Lync, users can keep track of their contacts’ availability; send an IM; start or join an audio, video, or web conference; or make a phone call—all through a consistent, familiar interface. Lync is built to fully integrate with Microsoft Office. The Microsoft Lync 2010 desktop client is available for Windows and for Mac and mobile versions are available for Windows Phone, iPhone/iPad, and Android devices.
via Business Grade Instant Messaging and Presence – Microsoft Lync.
Askozia
Askozia provides a highly intuitive telephone system for businesses. This is accomplished by combining our popular software AskoziaPBX with standard PC hardware or numerous embedded platforms. AskoziaPBX is incredibly easy to use, supports VoIP, ISDN, analog and GSM, is multilingual and costs less than a single VoIP phone.
via Askozia – Intuitive Telephony – Welcome!.
Call Flow Editor
A lot more than just an IVR tool. Completely integrated in Askozia’s web interface, you use drag-and-drop to build your own dial plans. Create highly sophisticated dial plans with Queues, If, Switch and Goto modules. Simply record announcements with your desk phone. Use the included templates if you don’t like to start from scratch. See more.
TiKL Touch Talk Walkie Talkie
Real Push-To-Talk + Chat/Text/Messenger. Supports 1:1 and groups. All for FREE
via TiKL Touch Talk Walkie Talkie – Android Apps on Google Play.
baresip – Open Source SIP User Agent
Baresip is a portable and modular SIP User-Agent with audio and video support.
via baresip – Open Source SIP User Agent.
Design goals
- Minimalistic and modular VoIP client
- SIP, SDP, RTP/RTCP, STUN/TURN/ICE
- IPv4 and IPv6 support
- RFC-compliancy
- Robust, fast, low footprint
- Portable C89 and C99 source code
PBXMate-FreeSWITCH-integration
The PBXMate software product from SoliCall is designed to improve voice quality by canceling echo, canceling noise and monitoring quality indicators. This article describes, in details, one option to integrating the PBXMate with FreeSWITCH in which both products are installed on the same Linux machine and a DNS is being used.
ÜberConference lets you manage conference calls visually
These are the problems that ÜberConference, a new (and currently free) conferencing tool from Firespotter Labs, was created to solve. Several of Firespotter’s employees come from the team that developed Google Voice, and their stated goal with ÜberConference is to develop a similarly innovative and useful tool for conferencing. We experimented with the tool to see how much it could streamline the teleconference experience.
First look: ÜberConference lets you manage conference calls visually | Ars Technica.
Once the conference has ended, you can view a small conference summary that will show you how long the call lasted, who talked the most, and who talked the least. If you elected to record the call, a small speaker icon will appear next to the call—click it to listen to the call in your browser window or download the MP3.
ÜberConference is still in the early stages of development, but at the moment it looks like an elegant (if not perfect) solution for many of teleconferencing’s shortcomings. I would still like to see ÜberConference let users assign their own conference numbers to ÜberConference accounts, but if you can get every member of your team to use the service, it provides useful visual feedback and features like call recording and call history that make it worth considering—even if you already have a conferencing solution.