The below features support when you do a SIP audio conferencing.
- SIP for signaling and RTP for media;
- Can support multiple simultaneous conferences;
- Audio mixing is done. Participants do not get their own audio;
- Support video replication. All participant with video capabilities can receive and send video. Hetrogeneous conferences with some audio terminals and some video terminals is also possible.
- Playout delay algorithm is implemented to create a synchronised audio stream;
- Works with sipc and other SIP user agents; H.323 users can connect using the siph323 software.
- Conferences can be setup and configured using a web based interface;
- Currently supports G.711 A law, Mu Law, G.721, DVI ADPCM, GSM and G.722 (high quality 16kHz) audio codecs;
- Can use both IPv4 and IPv6 for both session establishment (SIP) and media (RTP/RTCP). Allows heterogenity;
- Support for file sharing among participants;
- Restricted conference participation based of SIP digest authentication;
- Conference recording of audio streams. Sipconf can now write .au or .wav files for conference audio.
- Load balancing among multiple conference servers based on capacity of the conference server and maximum participant count of a conference.
- Modular design (components can be reused selectively);