I have to setup the 70 party conference bridge over VOIP. Can any one let me know what bandwidth I required to establish 70 party conference bridge at a same time. The conference will be more than 1 hour.
Can you please help me out, how to setup and which codec is the best for good sound quality as well how much bandwidth I need at the server side as well SIP user side.
Thanks in advance.
The best codec for calls AFAIK is G711 64 kbit/s bitrate (8 kHz sampling frequency x 8 bits per sample).
Setting it up, I don’t know i’m new to asterisk myself!
My gut feeling is that 70 parties will require specialist equipment, or at least VOX, preferably in the conference bridge, but at least in all the contributing phones. Otherwise I would expect horrific echo problems, or the need for significant attenuation in the bridge.
The actual bandwidth required is somewhat greater than the payload bandwidth.
Forget meetme for this.
We have a customer cluster for scaling above this limit, using app_conference.
as to bandwidth requirement thats simple
to you will need at least a 10meg pipe to avoid collisions and jitter
as unlike a system with 70 calls in progress where total bandwidth is required all the time as people all talk at different times, with conf server when one person speaks that goes to all parties.