----- torbenh@xxxxxx wrote: > On Sat, Mar 04, 2006 at 02:44:43AM +0900, hard off wrote: > > no, i mean like actual jamming..in realtime. > > > > still science fiction, yeah? i think it must be getting closer > > though. i have had some pretty good realtime phone conversations. > > don't get much lag over the phone line. > > > > of course sending actual audio data would be crazy, cos there's not > > even very good live audio streaming available yet....but simple > > control data must be possible, right? Kind of off topic, but the Asterisk PBX can send a 64kbit RTP stream in realtime with no latency. The current implementation is for voice only, so the ulaw codec is fixed at a mono stream at 8000hz. Not too practical for music. I would guess that it's not impossible to up the bitrate a little by hacking the source to allow for a mono 44100hz stream. The transport is UDP, therefore it'll simply drop packets that don't get there. There's also the issue of signalling. The current sane options are SIP and IAX2. SIP seems like overkill. I'd go with IAX2. I could imagine a situation where you run an IAX2 softphone with JACK inputs through a hacked Asterisk PBX. Science fiction? Not at all. Hard? Probably. -lee
