Web-SIP_phone_in_a_web-browser

Browser-based web telephone with the SIP support

Web Call Server supports audio and video calls from a browser to SIP devices, PBX servers, SIP-GSM gateways, VoIP conferences and various devices supporting the SIP protocol

Web application in the browser works as a softphone (a software SIP-powered telephone), accepts and initiates voice and video calls. If GSM or PSTN gateways are used, a voice call between the browser and a landline or mobile GSM phone can be established.

Specifications

Calling
platforms
Protocols
and technologies
SIP codecs SIP functions
  • Chrome
  • Firefox
  • Opera
  • Safari
  • IE
  • Edge
  • iOS SDK
  • Android SDK
  • SIP
  • WebRTC
  • Flash
  • RTMP
  • RTMFP
  • H.264
  • VP8
  • G.711
  • Speex
  • G.729
  • Opus
  • DTMF
  • Hold
  • Transfer
Calling platforms
  • Chrome
  • Firefox
  • Opera
  • Safari
  • IE
  • Edge
  • iOS SDK
  • Android SDK
Protocols and technologies
  • SIP
  • WebRTC
  • Flash
  • RTMP
  • RTMFP
SIP codecs
  • H.264
  • VP8
  • G.711
  • Speex
  • G.729
  • Opus
SIP functions
  • DTMF
  • Hold
  • Transfer

Web SIP telephone operation diagram

A customer uses the web browser with the connected microphone and headset for audio calls. Web Call Server bridges the gap between browsers and SIP devices and is in charge for exchange of audio and video data between the browser and the SIP part.

schema_SIP_video_phone_incoming_call_SIP_WCS_WebSocket_browser

 

Depending on SIP gateway capabilities, browser connection can be established with another SIP device, a mobile GSM phone or a landline phone. If the SIP gateway also supports video calls, the connection also provides video call capabilities.

Step-by-step diagram of a call between a browser-based web telephone and a SIP device or GSM phone

  1. The browser invokes the “call” function.
  2. The Web Call Server converts the call from the browser to the INVITE request readable by the SIP gateway.
  3. The SIP gateway calls the remote recipient.
  4. The callee answers the call.
  5. The SIP gateway sends the a “200 OK” SIP status response meaning the callee has answered.
  6. The Web Call Server returns the ESTABLISHED status, which means the connection is established successfully.
  7. A connection is established between the browser and the SIP device to transfer audio or video traffic. The traffic goes via the Web Call Server and optionally via the SIP Gateway (depends on the gateway settings).

 

callflow_SIP_video_phone_incoming_call_SIP_WCS_WebSocket_browser

 

Detailed information on testing a browser-based SIP phone is available on the Testing  page.

Download Web Call Server 5

System requirements: Linux x86_64, 1 core CPU, 2 Gb RAM, Java

    Download Now    

Installation:

  1. wget https://flashphoner.com/download-wcs5.2-server.tar.gz
  2. Unpack and install using 'install.sh'
  3. Launch server using command 'service webcallserver start'
  4. Open the web interface https://host:8444 and activate your license

 

If you are using Amazon EC2, you don't need to download anything.

Launch WCS5 on Amazon EC2

 

Web Call Server Monthly Subscription

$75 per month

 

   Purchase   

 

 


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