- Support for audio calls in Microsoft Edge. With Microsoft recently updating the browser’s web and audio support, Voximplant is one of the first companies to make its platform compatible with Edge. With Web SDK 4.0, web apps are empowered to handle incoming and outgoing voice calls from within the Edge web browser. Video calls in Edge will be functional as soon as VP9 or h.264 codec support is added to the browser.
- Video can be enabled during an existing audio call. Now developers can create web apps that offer a Skype-like “enable video” option during active audio calls. Video can be enabled or disabled at any time during the call.
- Both local and remote audio and video streams can be modified by applying filters. Developers can now modify video and audio streams in realtime. Possible applications of this feature include using video and audio filters to add watermarks, hide or alter faces, mask voices, etc.
- h.264 video codec can be set as “high priority”. The “h.264” video codec is known for its valuable speed in modern hardware, especially in mobile devices. Developers can now force this codec to provide better video quality with less CPU usage.
Additional enhancements include:
- Reduced latency of audio and video calls.
- Improved syncing between audio and video tracks.
- Calls within the Chrome browser now maintain better performance – even with poor network conditions.
- Audio and video codec priorities can be manually configured.
- Full support for WebRTC to offer users the best in voice and video quality.
New SDK also supports asynchronous initialization and drops support for Flash, which can still be used with legacy systems via 3.x versions of the SDK. Complete documentation for Voximplant’s Web SDK 4.0 beta can be found at https://voximplant.com/docs/references/websdk4/
One of the most popular scenarios for our Web/Mobile SDK is about enabling true click-to-call/tap-to-call function in web and mobile applications. True click-to-call means that it’s not about callback version when a call is initiated from the platform to both parties and connects them together, it means that call is initiated from the client side and goes via IP to the platform and then routed to required destination. True click-to-call is cheaper, faster and offers better UX; of course, you need to have the internet connection to use it (shouldn’t be an issue these days). This tutorial explains how to easily embed click-to-call into your app using VoxImplant SDKs and setup call routing using VoxEngine scenarios.
We recommend to check our Quickstart before you proceed
Full Featured Instant Messaging
Our first release of instant messaging functionality offered limited number of functions for 1-to-1 communication, we’ve been working on instant messaging upgrade that enables group messaging, messaging history, contact list control and other capabilities for full featured IM system. IM/presence should be explicitly enabled in VoxImplant application settings before it’s available.
We are working on the tutorial explaining how to use new functionality and will update our simple messenger application on GitHub in the near future. Other features for IM we have in the roadmap: search in history, checking presence from VoxEngine scenarios, sending messages from VoxEngine scenarios and via HTTP API.
New Instant Messaging functionality is only available in WebRTC mode
Web SDK improvements
VoxImplant Web SDK now supports Temasys WebRTC plugin for IE/Safari. We are working on ORTC support in Edge, but a lot of companies still use IE/Safari and want to have access to VoxImplant functionality in these browsers.
A lot of people use video conferencing functionality in Skype, but Skype is standalone application and it’s hard to integrate it with your own service. VoxImplant lets developers embed similar functionality into any web or mobile application. This tutorial will explain how to build video conferencing service with dial-in/dial-out functionality (to connect PSTN participants) and browser-based client application using WebRTC capabilities and VoxImplant.
Mozilla recently released Firefox 34 and there were some changes in WebRTC stack that weren’t compatible with our Web SDK. We have fixed most of them, p2p video calling will be fixed on Monday.
There are a lot of scenarios when you need to transfer a call to some other user or join it with another call by sending some command from a client built using Web SDK. New transferCall function enabled this functionality. It’s rather simple function – just provide two call instances as its parameters and VoxImplant will try to join call1 with call2. If you want to know the result of the transfer you can add the following event listeners to call1 – VoxImplant.CallEvents.TransferComplete and VoxImplant.CallEvents.TransferFailed.
If call was transferred successfully both call1 and call2 will be disconnected from Web SDK. In the world of telephony people usually call this function “Attended transfer”.
- Jan 19, 2017 08:38
- Speech-to-text: ASR
- Jan 19, 2017 08:38
- Speech-to-text: transcribing
- Jan 19, 2017 08:37
- Audio Recording
- Dec 28, 2016 12:47
- VoIP Push Notifications support for iOS SDK has arrived
- Dec 14, 2016 02:00
- New billing for video calls running via servers
- Dec 07, 2016 02:29
- Blind transfer support for SIP phones
- Dec 02, 2016 06:18
- Step-by-step call center tutorial part 8
- Nov 22, 2016 11:01
- Step-by-step call center tutorial part 7
- Nov 14, 2016 04:04
- Step-by-step call center tutorial part 6
- Nov 11, 2016 09:27
- High quality Speech Recognition is now available
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