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

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/

* 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

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

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

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

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 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.

hi everyone: does anybody know a way of shaping dhcp clients bandwidth? the only way of doing this that i know is using pppoe-server and limit the ppp interface, but it seems to be a little problematic protocol for me. im looking for a solution that doesnt require too many changes in the client too. my search in google led me to a strange white paper from juniper networks, introducing a

hi everybody. i have a bridge, and i want to apply QoS with htb and layer7 on both interfaces(eth0 and eth1), should i apply qdiscs and classes to each individual interface (eth0 and eth1, not br0)? if someone is using layer7, which is the right place to put the iptables rules to assure that all packets (fom internet to LAN and viceversa) get analyzed for layer7 patterns, including those

I have just finished setting up ices2 => icecast 2 on Suse9. Have 5 or so .ogg tracks in a playlist. Also downloaded a full copy of winamp 5 for win. If I point my browser at 10.1.1.1:8000/playlist.ogg then I get the screen to download file I dont know what is but winamp doesnot up. If I use the open URL feature in winamp and give it 10.1.1.101:8000/playlist.ogg Then winamp try to connects

Hi! I have playing my icecast using Ice for Vorbis and ezstream for mp3 , of course , one by one. Now my question I want to have a few channels with different kind of music. How can I do this? Can I have a few mount point into ices or ezstream or can I run a few instance with different config files? I try this with ezstream but it doesnot work. Any cluees ? Edwin Quijada

Hi! I am interesting to do a manager playlist using Perl for Ices0.3 I read that it is posible using Perl intead of Builtin in module clause. My question is how can I use different functions to play , next and other to ices. Somebody has done something like this? Where can I find info about this topic? I want to play my list from a database to change , modify and whatever I want my list. TIA

I try to do this. This was my first option but I get and error. Now I am using Ices0.3 to play mp3 . just one this list. When I try to do this I get this errror #>ices -c /usr/local/icecast/conf/conf2.xml & Error during send: Mount failed on http://10.0.1.1:8010/mp4, Error Couldnt connect Error during send: Mount failed on http://10.0.1.1:8010/mp4, Error Couldnt connect Error during

OK I thought change the port because I need a different port for another channel. Each mount must have a different port? Can u send me the config files for using 2 channels for seeing what are the differences. <p>>From: Geoff Shang <gshang@pacific.net.au> >Reply-To: icecast@xiph.org >To: icecast@xiph.org >Subject: Re: [icecast] Different play channeels >Date: Sun, 23

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

> 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.

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

Someone that know about FreeBSD!!! I need configure a DNS Server !!! Thank!!! _________________________________________________________________ MSN Amor: busca tu ? naranja http://latam.msn.com/amor/