sip audio

Hello,

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;
• multi-threaded;
• 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);

Thanks,
CSS_ETS03

hi friends ,,,
i am having doubt on sip audio conferencing ,,
while in this scenario which type of messaage header it can be support ,,,and which type of protocol it can b support