You can create new call, one side of call put in conference, other side of call put to sound card(create dummy one) or mixer. After that create streaming from that sound device, read stream in browser(use icecast).
Anyway i see 2 issues here:
I have tried that before (i am guru in asterisk) and not found way got delay <2seconds.
For setup all that need spend realy alot of time and looks like need be at least expert level in asterisk.
Other solution - use flash or java softphone for webpage and just call in automaticaly into conference. Need much less skill in asterisk, but need got that calling component(or pay for it) and need have average level in javascript.