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Web Call Server supports audio and video calls from a browser to SIP devices, PBX servers, SIP-GSM gates, VoIP conferences and other devices supporting the SIP protocol. Therefore, a web application can work in a browser as a software phone with the support for the SIP protocol, receive and initiate voice and video calls.
Overview
Supported platforms and browsers
Chrome | Firefox | Safari 11 | Edge | |
|---|---|---|---|---|
Windows | + | + | + | |
Mac OS | + | + | + | |
Android | + | + | ||
iOS | - | - | + |
Supported protocols
- WebRTC
- RTP
- SIP
Supported codecs
- H.264
- VP8
- G.711
- Speex
- G.729
- Opus
Supported SIP functions
- DTMF
- Holding a call
- Transferring a call
Management of SIP functions is performed with the REST API.
Operation flowchart
1: SIP server as a proxy server to transfer calls and RTP media
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- The browsers initiates a call using the /call/startup REST query.
- WCS connects to the SIP server.
- The SIP server connects to the SIP device receiving the call.
- The browser and the SIP device exchange audio and video streams.
Call flow
Below is the call flow when using the Phone example to create a call.
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12. Receiving a confirmation from the SIP server
Making an outgoing call from a browser to a SIP device
1. For the test we use:
- two SIP accounts;
- the Phone Video web application to make a call;
- a software phone to answer the call.
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7. To terminate the call, click the Hangup button in the browser or in the softphone.
Receiving an incoming call from a SIP device in a browser
1. For the test we use:
- two SIP accounts;
- a software phone to make the call.
- the Phone Video web application to answer the call.
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8. To terminate the call, click the Hangup button in the browser or the end call button in the softphone.
Known issues
1. It's impossible to make a SIP call if 'SIP Login' and 'SIP Authentification name' fields are incorrect
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