A guide for configuring SIP Video VoIP calling client for use with OpenVCX Video Conferencing Server. Read more
A guide for configuring and using WebRTC with OpenVCX Video Conferencing Server.
Using WebRTC and OpenVCX Video Conferencing Server
WebRTC is a SIP capable Real-Time Communications video chat client which runs natively in the browser. WebRTCP operates in Chrome and Firefox.
OpenVCX (Video Conferencing Server) supports WebRTC as a video client endpoint and video conference participant.
OpenVCX can be used to bridge video calls between WebRTC clients and other SIP video endpoints. OpenVCX also supports WebRTC endpoints in a multi-party conference together with any other supported SIP video endpoint.
The WebRTC Demo page is included in the OpenVCX installation package and can be accessible at the following url
WebRTC provides a means for easy and secure video calling. All media communications between WebRTC and the conferencing server is encrypted using SRTP.
WebRTC uses the websockets transport protocol for SIP (Session Initiation Protocol) signalling exchange. The OpenVCX websockets listener is typically two ports higher than the standard UDP/TCP SIP listener port.
Starting the Video Conferencing Server
Listening for SIP requests on 0.0.0.0:5060 udp
Listening for SIP requests on 0.0.0.0:5060 tcp
Listening for SIP requests on 0.0.0.0:5062 ws
Listening for HTTP requests on port 8090 ( http://127.0.0.1:8090/webrtc-demo/ )
In this case the WebRTC Outbound Proxy configuration should be set to ws://[ OPENVCX IP / HOST ]:5062.
STUN (Session Traversal Utilities for NAT)
A valid STUN server should be set when making video calls from within a NAT (Network Address Translation) to an OpenVCX server located outside of the local NAT, such as on the internet. The STUN server can be any public STUN server listening on STUN port 3478. A STUN server should not be utilized when OpenVCX is deployed on the same local network as WebRTC.
Using the default configuration OpenVCX does not act as a SIP registrar and can work with any WebRTC provided SIP username and domain. WebRTC will need to first register with OpenVCX prior to making a video call.
The WebRTCP SIP registration password will only be validated when using a conference SIP URI with a configured SIP password. This password must match any conference specific configured SIP password.
Dialing into a conference
To dial into a video conference the contact SIP URI should be set to a matching conference definition file residing on the OpenVCX server. For example to call the sample demo conference defined in the file VCX_INSTALL_DIR/conferences/1234.conference you would use 1234 as the username portion of the SIP URI.