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Web Call Server Usage
Web Call Server For
Site Owners
For your web project, you can add capability of audio and video calls from your website to mobile GSM and SIP devices. Use the power of Flash and WebRTC technology for browser calls.
Enterprise projects
Embed webphone into your existing CRM or helpdesk system. Users of your system will be able to call mobile GSM and VoIP devices directly from their web interface. The web phone uses the latest WebRTC technology for high quality browser calls and Flash technology to cover most popular browsers.
VoIP providers
Add a webphone to user’s front page and user will have an alternative webcall software that does not require any installation. Your users will call over webphone using the WebRTC and Flash VoIP capabilities of a web browser.
Audio video startups
Stay mainstream with us. Use the latest and most powerful WebRTC and reliable Flash as a platform for your web calls. The Flashphoner platform is perfectly suitable for creation of low-latency realtime applications such as web conference, video chat, streaming, VoIP, web messengers and so on.
Questions and Answers
What is REMB?
What is WebRTC NACK?
WebRTC NACK is Negative Acknow…
iPhone Safari WebRTC
Does iPhone Safari browser sup…
How to check all needed ports enabled for Web Call Server?
Web Call Server requires a set…
How to setup Click2Call for two different buttons, different SIP servers and directions?
I’m wondering how to setup 2 C…
Should the client side of Web Call Server be located on Linux server? Can we deploy it on Windows server?
The client side can be located…
Does WCS support multiline WebRTC SIP calls?
For example, if we have a web …
What does this mean “SIP/GSM” calls. Do you provide GSM calls?
No, we are not a GSM provider. Bu…
Is there any SDK or API which can help to integrate it with our existing flash application?
We provide open sourc…
Is there any connection limitation for Wb Call Server with a valid purchased license?
If you have a valid WCS licens…
Major improvements and Bugfixes
WCS 5.1 Changelog Q1-Q2 2018
16 August 2018
WCS 5.0 Changelog Q1-Q2 2018
16 August 2018
WCS key releases 2017, August – Dec
14 November 2017
WCS key releases 2017, May – July
31 July 2017
WCS5 release notes Jan – April 2017
2 May 2017
navigator.mozGetUserMedia has been replaced by navigator.mediaDevices.getUserMedia in Firefox 45
25 April 2016
We Have Implemented Simple Administrator Interface
18 March 2016
We Have Added Powerful Monitoring Add-on For WCS3
21 October 2015
News And Articles
Sales migration from Skype to Teams on 16 Apr 2025
21 April 2025
Card payments were successfully restored on September 13, 2024
13 September 2024
Card payments temporary unavailable since August 16, 2024
16 August 2024
Initiating a WebRTC server on Windows 10 using WSL2
19 October 2021
How to use Docker with WebRTC in production
1 July 2021
WebRTC Stream Recording Load Test
20 May 2021
WebRTC Back and Forth
17 May 2021
How many mixers can a server have?
13 April 2021
Comparing load testing methods
22 March 2021
You add an HTML5-code into your project
or use our mobile SDK
WCS ensures streaming
and web calls
WCS integrates with your web-server
over REST / HTTP
Streaming Video
Web Call Server supports all popular streaming video web-technologies such us WebRTC, Flash, RTMP, RTMFP, RTSP, SIP, and Websocket streaming which allows to deliver a video stream to a wide range of browsers and mobile devices
Add an RTSP-player into a web-page or mobile app
Playback platforms
- Chrome
- Firefox
- Opera
- Safari
- Safari, iOS
- IE
- Edge
- iOS SDK
- Firefox
- Opera
- Safari
- Safari, iOS
- IE
- Edge
- iOS SDK
Codecs
- H.264
- VP8
- AAC
- Speex
- G.711
- VP8
- AAC
- Speex
- G.711
Technology
- WebRTC
- Flash
- Websocket
- Flash
- Websocket
Video stream from a web-browser or mobile app will be recorded on the server-side and available for downloading and further processing
Formats
- webM
- mp4
- mp4
Technology
- WebRTC
- Flash
- Flash
Codecs
- H.264
- VP8
- Vorbis
- Speex
- G.711
- VP8
- Vorbis
- Speex
- G.711
Two-way WebRTC video chat with low-latency in browsers and mobile devices
Platforms
- Chrome
- Firefox
- Opera
- Safari
- IE
- Edge
- iOS SDK
- Firefox
- Opera
- Safari
- IE
- Edge
- iOS SDK
Technology
- WebRTC
- Flash
- Flash
Broadcasting from a video capturing device or software over RTMP protocol and low-latency delivery to browsers and mobile apps
Platforms
- Chrome
- Firefox
- Opera
- Safari
- Safari, iOS
- IE
- Edge
- iOS SDK
- Firefox
- Opera
- Safari
- Safari, iOS
- IE
- Edge
- iOS SDK
Technology
- WebRTC
- Flash
- Websocket
- Flash
- Websocket
Encoder protocols
- RTMP
Encoder codecs
- H.264
- AAC
- AAC
Web - SIP calls
Embedding of a browser based web phone into a web page or mobile app, full integration with SIP, calls to mobile and PSTN phones via a SIP-gateway, receiving inbound calls in a browser
Web phone on your website for online calls to SIP and mobile phones via a SIP gateway
Platforms
- Chrome
- Firefox
- Opera
- Safari
- IE
- Edge
- iOS SDK
- Firefox
- Opera
- Safari
- IE
- Edge
- iOS SDK
SIP features
- DTMF
Technology
- WebRTC
- Flash
- Flash
Codecs
- G.711
- Speex
- Opus
- Speex
- Opus
Resources
All needed resources for development and integration
Android SDK
- Coming soon ...
Try Web Call Server
This is 30-days trial version. Please fill out the form to get license.
Docs
User Guide
Admin Guide
Developer Guide
Call Flow
Specification
Web Call Server Installation
System requirements: Linux x86_64, 1 core CPU, 1 Gb RAM, Java
1. Download Web Call Server 5
2. Install using the 'install.sh' script
3. Launch using 'service webcallserver start'
4. Open the web-interface http://host:9091 and activate your license
5. To get this working with WebRTC you have to import SSL certificates and open https://domain:8888
Launch on Amazon EC2 does not require installation
