Open Source Video Conferencing and Streaming Server


Quick start installation guide for OpenVCX Video Conferencing Server.  Read more


Basic Startup Configuration

Quick start configuration guide for OpenVCX Video Conferencing Server.  Read more


SIP Video Client Configuration Checklist

A guide for configuring SIP Video VoIP calling client for use with OpenVCX Video Conferencing Server.


Using WebRTC

A guide for configuring and using WebRTC with OpenVCX Video Conferencing Server.  Read more


SIP Video Client Configuration for use with OpenVCX Video Conferencing Server


The OpenVCX Video Conferencing is designed to be used with standard SIP VoIP (Voice over IP) and Video VoIP calling clients. Some examples are below.

  • Bria
  • Linphone
  • Vippie
  • WebRTC
  • X-Lite

Use the guidelines below when setting up a video calling client.


SIP Connectivity

If using OpenVCX in stand-alone mode, a client will first need to register in-order to succesfully place a video call. In this case you will need to set the address of the OpenVCX server as the SIP Proxy.

For eg., SIP Proxy:

The client SIP account can use any user id, password, and domain.

When dialing into a conference which restricts which participants may join, the SIP user account name should match one found in the allowedUsers property list of the conference definition file.

OpenVCX does not support SIP presence as it fulfills the role of a conferencing media server.


Video and Audio Codecs

To succesfully establish video communication, the client and OpenVCX will have to agree on a common media codec available in the SDP offer. Please refer to the list of media codecs supported by OpenVCX. A matching codec and encoding profile needs to exist for the client to be succesfully connected.

Each conference definition file may contain codec specific encoding properties. The client must support at least one video encoder profile to establish video communication.

If no video SDP offer candidates can be fulfilled by the server, the media communication will be established as audio-only. OpenVCX may also support an SDP offer which does not contain video to allow a participant to join a conference as an audio-only participant.


STUN (Session Traversal Utilities for NAT)

STUN should be enabled on the client for it to discover it's own public IP. The client SDP offer will contain it's discovered public IP in the connection address field. STUN may be disabled when testing in a local area network.

The server will respond to STUN binding requests and will issue it's own binding requests for the purpose of connectivity establishment.

The client may use a public STUN server included in its configuration.


ICE (Interactive Connectivity Establishment)

If possible, ICE should be enabled on the client. OpenVCX will try to choose the best ICE candidate from the client SDP offer.

TURN (Traversal Using Relay NAT) can be disabled since the client phone will always establish a media connection with OpenVCX . In this way, OpenVCX behaves like a relay server and should not be deployed behind a NAT.


DTMF (Dual-tone multi-frequency) signaling

It is preferred that the client send any DTMF tones using either SIP INFO or DTMF 2833, not in-band with the audio media.







Don't hesitate to contact us via our contact page or email us at