similar to: How calculate bandwith - How listeners

Hi! I am newbie using broadcast for internet. I have two lines ADSL 768Kbps each one. Each line has connected a server doing stremaing with icecast. Somebody tell me that my bandwith is so low for much listeners. He told me that max people listen my radio station is 32 :( I wanna know how if it is true. I have a project to build my own radio station using my lines but this way it is imposible.

* Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > I use kmix as mixer, but there is nothing in it about a > > "capture channel". How could I find where it is defined ? > then try alsamixer or amixer and watch out for capture and unmute and apply > this for the according channel (mic, line-in, pcm or master) Hum... the problem is that RH 9 uses

* Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > No "R" on the pcm line, so that is probably the problem : my soundcard can > > only record line1... > doesn't this mean that pcm is captured at the moment? > just try this: use aumix, go to mic, pcm or master and press space, then > this might be captured (a R appears left to the bar). I

wow, Enrico ... it's a great help ;-) I comment: ----- Original Message ----- From: "Enrico Minack" <enrico.minack@informatik.tu-chemnitz.de> To: <icecast@xiph.org> Sent: Wednesday, February 11, 2004 8:00 PM Subject: Re: [icecast] Q: Is it possible? <p>> Hi Raúl, > > interesting project ;-) So this is what I would recommend. For a number of > remote

On Wednesday 24 March 2004 03:53, Enrico Minack wrote: > Why do you consider livecaster's stream being more efficient than the > HTTP-Stream? Actually, after the HTTP-Header there are just raw MP3-Data. > In comparision to that, livecaster puts these MP3-Data into an > RTP-protokoll, which produces more overhead than 'raw' http. And you may be > faced random packet loss.

What SoftPhone working very well with *? S.O. is Debian Linux Thanks for your comments. JRR _________________________________________________________________ MSN Amor: busca tu ? naranja http://latam.msn.com/amor/

> You will hit two problems though. The first is that you will need to have > a stream for remote participants to listen to. Yeah, that's right. If you want the remote speakers to be able to listen to the other speakers this becomes a little complicated. If you hear your own voice with a latency more than 1/10 or 1/5 second it becomes very distracting! This latency really is a problem.

Hello, I am looking for a documentation of the communication protocolls icecast is capable of: ICY, XAUDIOCAST and a modified HTTP. In order to write client software being capable of all available features I was searching on the net for those documentations but no luck so far. How come? Isn't this protocoll documented? How does the developer team know what options and functins are available

> > I am looking for a documentation of the communication protocolls icecast is > > capable of: ICY, XAUDIOCAST and a modified HTTP... > You should use libshout2. It's a handy dandy library... Yeah, I am familiar with this library, but as I know this is just for sources. I am more interested in requirements clients have to meet, what header response options are available, and

Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job

> What are my options for grabbing a source at one server and sending > it on another without re-encoding? find a client for icecast (mpg123, wget, fetch, ...), put the content to standard out, find a source for darwin (sorry, I don't know any) and let it use the standard in. Enrico --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage:

I just installed Icecast2 and the installation was a sucess. However, when I go to startup Icecast. It's reply is: [root@linuxserver bin]# ./icecast -c /etc/icecast.xml WARNING: You should not run icecast2 as root Use the changeowner directive in the config file [root@linuxserver bin]# <p>So then I go into the icecast.xml and alter the changeowner User to admin. Here is a bit from my

hi! I'm using the speex library in my RTP project (i'm using GNU ccRTP library). My program creates and destroys speex objects everytime a new RTP connection is made. But, when i test it with a client which create a connection to it one after another, the free memory decrease constantly to the point where the programs execution crash because not more free memory. It seems like the

Thanks Geoff, it's becoming more clear to me now... > So, assuming it does, you could try: > aumix -w r [yann@raglou yann]$ aumix -w R [yann@raglou yann]$ aumix -q vol 100, 100 pcm 100, 100 speaker 0, 0 line 0, 0, P mic 4, 0, P cd 0, 0, P igain 0, 0, P line1 0, 0, R phin 0, 0, P phout 0, 0 video 0, 0, P No "R" on the pcm line, so that is probably the problem : my soundcard

* EvilOverlord (eviloverlord@kucs.net) écrivait : > In whatever you use as mixer control for the soundcard (alsamixer, > aumix, etc) what is set as the "capture" channel? Your soundcard may > not support capturing what is being played. If it helps : my soundcard is a ES1988 Allegro-1, the module is "maestro3". I use kmix as mixer, but there is nothing in it about a

Hola estoy tratando de configurar mi primera VPN, pero cuando me conecto al servidor VPN Netstat -nat me dice que la coneccion esta en estado TIME_WAIT, por otro lado revisando syslog encuentro lo siguiente: tincd 1.0.2 (Nov 8 2003 20:54:15) starting, debug level 0 Nov 22 08:42:15 woody tinc.vpn[5810]: /dev/net/tun is a Linux tun/tap device (tun mode) Nov 22 08:42:15 woody tinc.vpn[5810]:

On Wednesday 24 March 2004 11:00, Enrico Minack wrote: > > ...HTTP is very inefficient compared to RTP. > > what exactly do you mean with 'efficient'? Used bandwidth or available > features? In terms of used bandwidth, RTP is the clear winner. In terms of features, neither is a clear winner: HTTP has features RTP doesn't have, RTP has features HTTP doesn't have.

On Sat, 2004-03-27 at 07:23, Pierre Lazuly wrote: > * Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > > I use kmix as mixer, but there is nothing in it about a > > > "capture channel". How could I find where it is defined ? > > then try alsamixer or amixer and watch out for capture and unmute and apply > > this for the according

Hi Raul, Here is how I would set it up to answer your scenarios: -One Icecast server in Spain. -One source in Spain -One source in Honduras -One source in Guatemala (for example) ... I would reserve a mountpoint for each one of these sources: -Spain = my_radio.mp3 (Your listeners will listen to this mountpoint ONLY) -Honduras = Honduras.mp3 -Guatemala = Guatemala.mp3 ... Let's say you use

Hi from spain! (sorry, I speak only a little english) I collaborate with an spanish NGO and we're planning to set up an Internet radio station. I would like to base our project on linux and Icecast / liveice (right?) but I don't understand some terms of "radio" practices. I need your experience&help; the project should cover the following scenarios: * A broadcast server