similar to: users.map file

With some carriers the initial Audio (2-4 secs) seems to get cut off when using a Auto Attendant or Conf Meetme. Is there any known remedies for that. Just want to know if others have seen that esp. with Level 3. If Auto Attendant says - "Welcome to ABC bank" Caller only hears "Bank" -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks for this information, but the two connections do not die out when using my player in the Android browser. In fact they even stay when the Audio.src is set to some invalid string. You can observe this here: http://concertzender.nl/wp-content/themes/cz-theme/App/test/ , by clicking one of the large icons - and then monitor the corresponding stream here: http://streams.greenhost.nl:8080/

hello, now that html5 audio becomes usable in some browsers, for example http://webaudiodemos.appspot.com/AudioRecorder/index.html ... has anybody thought about to implement a web based source client for icecast? that's for now beyond my coding skills, unfortunately... thx, uno

I’m setting up my system on a Mac, and have successfully gotten Ladiocast to feed Icecast and can access it on my server Mac at localhost:8000/stream. If I try to access it at the IP of the Mac at 192.168.0.5:8000 or port forwarded from the internet, I CAN successfully access the admin page. However, if I try to access the stream from either the local IP or port forwarded from the internet, I

It's behaving as it is meant to: if a listen client gets too far behind, Icecast2 server is kicking them off. error.log says, "INFO source/source.c Client 120 (xxx.xxx.xxx.xxx) has fallen too far behind, removing" I don't see a setting in icecast.xml that sets the value for "too far behind". I'm guessing it's related to <queue-size>. I wish Icecast

Five years after initially posting to this list[0], I finally completed as browser-based video source client iceast. The code is here: https://gitlab.com/jamie/icecream As Romain Beauxis responded to my initial email, webrtc was unsuitable and websockets was the way to go. Thanks for the help! jamie 0. http://lists.xiph.org/pipermail/icecast-dev/2013-July/002223.html -- May First/People Link

Dear all, The problem I am facing, is that one of my clients is concerned about data usage. My client monitors the current number of listeners in the Mount Point list streams.greenhost.nl:8080. The case is, when I start a stream from an HTML app on an Android device, I can see two listeners being added to "Current Listeners". This peculiar behaviour only occurs on Android, not on

Thanks, Brad -- I first turned to VLC but my stream-client device is a Raspberry Pi 3, and there doesn't seem to be a build for the device. That said, I just stumbled across mpg123, a command-line mp3/stream player that has audio buffer and timeout setting which so far looks quite robust. For Icecast2 content, when Icecast kicks a client off for being "too far behind" -- what

2014-05-03 18:40 GMT+02:00 Brad Isbell <brad at musatcha.com>: > The general idea was to make a really simple source client that could be > launched with a single click (such as from a hosting provider's admin > panel). The client accepts parameters on the query-string for configuring > so that in a real-world environment, the user would not have to configure > anything.

On Sun, Oct 21, 2012 at 11:01:37AM -0700, Tobias Grosser wrote: > On 10/21/2012 10:57 AM, Jack Howarth wrote: >> On Sun, Oct 21, 2012 at 10:38:48AM -0700, Tobias Grosser wrote: >>> On 10/21/2012 09:13 AM, Jack Howarth wrote: >>>> On Sun, Oct 21, 2012 at 08:38:21AM -0700, Tobias Grosser wrote: >>>>> On 10/20/2012 05:38 PM, Jack Howarth wrote:

Brad Isbell: > I have built a web-based source client using the Web Audio API and > You can see a demo here: http://demo.audiopump.co:3000/ oh my, that looks great!!! > Let me know if there is interest. I can easily add the native Icecast > compatibility. definitely! i've only an icecast server available... thx, uno

I looked at these requests and started trying to interpret the data of the header and the stream. I receive a header, and even when I added the parameter to the request to try and get meta-data, it seems that it is not being included. Namely, the number that is supposed to tell the distance between each meta-data section that is sent in the stream. I couldn't figure out the config setting

Thomas, Do you by chance know which players are ignoring the content-type headers? I have done quite a bit of testing for that exact issue but haven't ran into any problems. Perhaps I am testing the wrong players. Thanks, Brad Isbell brad at musatcha.com http://www.musatcha.com On Fri, Nov 21, 2014 at 7:49 AM, "Thomas B. R?cker" <thomas at ruecker.fi> wrote: > On

On Sun, Oct 21, 2012 at 02:35:49PM -0400, Jack Howarth wrote: > On Sun, Oct 21, 2012 at 11:01:37AM -0700, Tobias Grosser wrote: > > On 10/21/2012 10:57 AM, Jack Howarth wrote: > >> On Sun, Oct 21, 2012 at 10:38:48AM -0700, Tobias Grosser wrote: > >>> On 10/21/2012 09:13 AM, Jack Howarth wrote: > >>>> On Sun, Oct 21, 2012 at 08:38:21AM -0700, Tobias

"Live streams are not live, they can be up to 15 to 20 sec off original encoding." Something you become aware of very quickly when you're announcing into a mic and have to wait to hear your words! We do a bit of pre-show calibration with folks at the station to determine how long the delay ("latency" in digital-speak) is at the start of the remote broadcast, and start

On 05/03/2014 04:40 PM, Brad Isbell wrote: > I have built a web-based source client using the Web Audio API and > getUserMedia, called the AudioPump Web Encoder. It uses the SHOUTcast > source protocol for now, but I have the code nearly ready for the > Icecast source protocol for when someone needs it. > > You can see a demo here: http://demo.audiopump.co:3000/ That's

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac

Hi there, Any other versions of Google Chrome and all version of other browsers are working. I have done my best to set up the intro to match the live stream. http://185.139.168.34:8000/yleisohjelma - intro is 256 kbits/s 48 kHz ISO-MPEG2 L3 - live is 256 kbits/s 48 kHz ISO-MPEG2 L3 http://185.139.168.34:8000/vara - live is 128 kBits/s 48 kHz, possible ISO-MPEG2 L3 - there is no intro Of

i was wandering if anyone could give me a free mountpoint to set up my little station for streaming? i will be streaming at 32 KBPS .AAC thanks Timothy Your friend in the music industry follow me on twitter @timothyclark13 to check out music, photos, videos, press, show dates, and more go to http://www.reverbnation.com/timothyclark13 feel free to give me a call at 7244011224 note, if

Logs indicate everything's running correctly, my player says I'm connected, and the Web page for the server says it's playing files, but there's no audio. Strange. Any thoughts? Did I possibly convert the MP3's to Ogg Vorbis incorrectly?