openj-gate.com

lechoixdeslibraires.com

open4u.co.uk

argosnear.me

sarf3omlat.com

opencities.ca

australia-opening-times.com

Web Call Server 5

Add streaming video or web telephony features to an existing
web or mobile app

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 web technologies for streaming video, such as WebRTC, HLS, RTMP, RTSP, SIP, and WebSocket streaming, which allows delivering a video stream to a wide range of browsers and mobile devices

HTML5-RTSP player for IP cams
Add an RTSP-player into a web-page or mobile app
Learn More
Device
IP cam
RTSP server
Publish
RTSP
- H.264
- AAC
- G.711
Features
TCP
UDP
REST API
Playback
Browser
iOS SDK
Android SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
CDN for low latency WebRTC streaming
Dynamic CDN for low latency WebRTC streaming
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Features
ACL
Transcoding
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
WebRTC stream recording
The video stream from a web browser or mobile app is recorded on the server-side and becomes available for downloading and further processing
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Features
Web API
REST API
Playback
MP4
- H.264
- AAC

WebM
- VP8
- Vorbis
WebRTC screen sharing
Screen sharing from web browsers using the WebRTC technology
Learn more
Device
Chrome
Firefox
Android SDK
Publish
WebRTC
- H.264
- Opus
Capture
Desktop
Window
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
WebRTC video chat
Two-way low-latency WebRTC video chat in browsers and mobile devices
Learn more
Participant
Browser
Android SDK
iOS SDK
Relay
WebRTC
- H.264
- Opus
Features
Room API
MCU
Custom impl
Participant
Browser
Android SDK
iOS SDK
Live broadcasting from a webcam
Online broadcasting from browser's web camera or a mobile app to other browsers and devices
Learn more
Device
Browser
Android SDK
iOS SDK
Publish
WebRTC
- H.264
- Opus
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
Streaming from an RTMP Live Encoder
Broadcasting from a video capturing device or software over the RTMP protocol and low-latency delivery to browsers and mobile apps
Learn more
Device
ffmpeg
OBS
RTMP Encoder
Publish
RTMP
- H.264
- AAC
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
WebRTC to HLS broadcasting
Live broadcasting to TV, set-top boxes and other devices via HLS
Learn more
Device
Browser
Android SDK
iOS SDK
Publish
WebRTC
- H.264
- Opus
Playback
HLS
- H.264
- AAC
HLS players
HTML5 video
hls.js
video.js
Multipoint Conference Unit
Each MCU participant plays only one mosaic stream
Learn more
Device
Browser
Android SDK
iOS SDK
Publish
WebRTC
Playback
Playback
Browser
Android SDK
iOS SDK
WebRTC video conferences
Live video conferencing for 3 and more participants
Learn more
Participant
Browser
Android SDK
iOS SDK
Relay
WebRTC
- H.264
- Opus
Features
Room API
MCU
Custom impl
Participant
Browser
Android SDK
iOS SDK
WebRTC as RTMP republishing
Broadcast WebRTC stream as RTMP to any live service or RTMP server
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Republish
RTMP, RTMPS
- H.264
- AAC
Services
Youtube
Facebook
RTMP servers
Player for a browser or mobile app
Universal player for playing video streams in a browser
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Playback
Browser
- Player
- Embed player
Android SDK
iOS SDK
Video stream mixing
Combining multiple video streams into one by using WCS
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Features
Grid
MCU
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
VR 360 streaming
Broadcast VR 360 stream
Learn more
Device
Browser
RTMP encoder
Publish
WebRTC
- H.264
- Opus

RTMP
- H.264
- AAC
Playback
VR player
- WebRTC
- RTMP
- HLS
- RTSP
PNG stream snapshots slicing
The server can do snapshots of broadcast in .png format on REST request
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Features
File system
Web API
REST API
Format
PNG
Stream transcoding
Change audio/video codecs, formats and bitrate of streams automatically and by REST request
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Features
Codec
Resolution
Bitrate
FPS
GOP
Quality
Profile
Level
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
Capturing VOD from a file
WCS allows you to work with previously recorded broadcasts as with a regular video stream
Learn more
Device
File system
AWS S3
Publish
MP4 file
- H.264
- AAC
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
Playing a video stream from the server in a browser via MSE
MSE technology is used when playing a video stream on a page if there are no strict requirements for delays
Learn more
Device
Browser
Android SDK
iOS SDK
RTMP encoder
IP cam
MP4 movie
SIP endpoint
Publish
WebRTC
RTMP
RTSP
VOD
SIP
Playback
Browser
- Player
- Embed player
Publication and playing parameter managing
Configuring the camera and microphone in the browser, managing various options for publishing and playing video and audio
Learn more
Browser
Chrome
Firefox
Safari
Edge
Device
Microphone
Camera
Screen
Speaker
Publish
Audio
- Mic Gain
- Mono/Stereo
- FEC

Video
- Width
- Height
- FPS
- Bitrate
- Protocol
- Codecs
Playback
Audio
- Volume
- Voice Activity

Video
- Width
- Height
- Bitrate
- Protocol
- Quality
- Codecs
Web-SIP phone in a web browser

Embedding 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-SIP phone in a web browser
Web phone on your website for online calls to SIP and mobile phones via a SIP gateway
Learn more
Participant
Browser
Android SDK
iOS SDK
Relay
WebRTC
SIP/RTP
Features
DTMF
Hold
Transfer
Call recording
Participant
SIP endpoint
- H.264
- VP8
- Opus
- G.711
- G.722
- G.729
- Speex16
Click to Call button
Online calls from a website to SIP and mobile / PSTN phones via a SIP gateway
Learn more Call
Participant
Browser
Android SDK
iOS SDK
Relay
WebRTC
SIP/RTP
Features
DTMF
Hold
Transfer
Call recording
Participant
SIP endpoint
- Opus
- G.711
- G.722
- G.729
- Speex16
Broadcast SIP call to a stream or RTMP
Broadcasting of a SIP video call to an RTMP server or CDN
Learn more
Device
SIP endpoint
- H.264
- Opus
- G.711
- G.722
- G.729
- Speex16
Publish
SIP/RTP
- H.264
- VP8
- Opus
- G.711
- G.722
- G.729
- Speex16
Features
DTMF
Call recording
Playback
Browser
Android SDK
iOS SDK
Player
- WebRTC
- RTMP
- HLS
- RTSP
Republish
RTMP, RTMPS
- H.264
- AAC
Services
Youtube
Facebook
RTMP servers
Resources

All needed resources for development and integration

Web SDK
Android SDK
iOS SDK
Download
Demo

 

Solutions

Ready-to-use images for one-click WCS deployment

WCS on Amazon EC2
WCS on DigitalOcean
WCS in Docker

 

Documentation

Detailed documentation for server, SDK and REST API

WCS
SDK
API

 

Web Call Server Installation

Minimal system requirements: Linux x86_64, 1 core CPU, 2 GB RAM

1. Download Web Call Server 5.2

2. Install using the 'install.sh' script

3. Launch using 'service webcallserver start'

4. Open the web-interface https://host:8444 and activate your license

 

Launch on Amazon EC2 or DigitalOcean does not require installation

 

Blog
Questions and Answers
What is REMB?

REMB stands for Receiver Estim…

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
H.264 codec is enabled by default for Web SDK, Android SDK, iOS SDK for WebRTC
9 January 2017
WCS server performance has been improved for WebRTC and RTMP streams
9 January 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
How many RTSP cams can I connect to the WebRTC server?
31 May 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