similar to: Relay/forward RTP-packets over icecast2

Hello to everybody,   I seem to be in struggle with the websockets in R. I wanted to download the streaming data from the BitCoin exchange MtGox directly to R, but R cannot establish the connection. The websocket specs are defined as: * Host: websocket.mtgox.com or socketio.mtgox.com * Port: 80 or 443 ( ssl ) * Namespace: /mtgox (Including beginning slash) url for more

Following up on this topic ( sorry if this starts a new thread but I just joined the ml ), I do no understand why it is not possible to use the audio stream from webRTC's getUserMedia and then send it over a websocket ? It seems that the webRTC implementation can natively encode in ogg format in stereo from any interface ( according to

I'm using rsync 2.63 on a NetWare 6.5 server backing up various volumes to a SLES 9 server. My script that I'm using on the NW server is: # Rsync synchronisation of APPS rsync -rRutzvP --volume=apps: ./ 192.168.1.252::SCA/apps # Rsync synchronisation of DATA rsync -rRutzvP --volume=Data: ./ 192.168.1.252::SCA/Data # Rsync synchronisation ofGWMAIL rsync -rRutzvP --volume=GWMAIL: ./

Hey all, So we have been advised from this thread https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 to not use http put as it is not in real-time, instead they are suggesting the use of SDP, is that something that icecast supports? Or does anyone have other ideas on this? ~stephen On Sun 12 May 2013 01:51:31 AM CDT, Thomas Ruecker wrote: > Hi, > > On 11

Hi, On 07/23/2013 07:44 PM, Jamie McClelland wrote: > I'm following up on a thread started by Stephen a couple months ago > about building a JavaScript source client using webrtc. > > The first step suggested was to figure out how to mux the audio and > video. After I posted a feature request on the webrtc experiment js > library, we seem to have a solution: >

I'm following up on a thread started by Stephen a couple months ago about building a JavaScript source client using webrtc. The first step suggested was to figure out how to mux the audio and video. After I posted a feature request on the webrtc experiment js library, we seem to have a solution: https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-20791759 Based on the last

Hi Stephen, > So we have been advised from this thread > https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 > to not use http put as it is not in real-time, instead they are > suggesting the use of SDP, is that something that icecast supports? Or > does anyone have other ideas on this? The imminent Airtime 2.4.0 release has support for Opus, and it

Hi Jamie, The webRTC API does not sound suitable for source->server streaming for many reason. For instance, the peer-to-peer connection requires input from both end and seems quite unfeasible to implement in a server. Likewise, codecs are completely abstracted and much more. In reality, webRTC is an API to acheive full-duplex conversations a-la skype and not for streaming. For these

involved. This is easy enough to verify, dump to a file as described on that page and feed the resulting file to 'ogginfo'. If it is indeed properly muxed into an Ogg container, then just forwarding it via HTTP PUT instead of writing it to a file should do the job. I'd suggest someone just tries it. If you need a Icecast server to test against I can also provide that. Ping me on IRC

Unfortunately, Dennis, my source stream is 128kbps and will never go higher. Is Liquidsoap still a good idea? On Fri, 04 Mar 2016 14:48:41 +0100, you wrote: >Great tool to do this: liquidsoap > >Keep in mind that transcoding degrades the quality of tour stream dramaticly. You can avoid this by feeding liquidsoap or stream transcoder with a highquality or even transparant stream and

Great tool to do this: liquidsoap Keep in mind that transcoding degrades the quality of tour stream dramaticly. You can avoid this by feeding liquidsoap or stream transcoder with a highquality or even transparant stream and trancode this to the different streaming formats you like. I used to do this by feeding a flac stream to liquidsoap and transcode this to 5 different stream formats i needed.

> Hi, > > On 01/24/2014 04:39 PM, geekshabeka at riseup.net wrote: >> How can I configure vp8 encoder in icecast 2.4 version? > > Icecast doesn't contain any encoders, at all. It just passes through > streams and deals with them on a container level (Ogg, WebM/MKV). ok, thanks, I was mixing gstreamer language with icecast language :-) > >> The normal way

Thomas, Thank you for your interest in this, you description is as accurate as I can see. > From my perspective your challenges will be to get the containers right. > WebM for audio+video > Ogg for audio > > Also (I'm not that familiar with webRTC) you might need to reencode > to Opus and VP8 in some cases? here is the great news

Hello list. I'm going te explain my trouble. I have my asterisk with a TDM400P with 4 FXS channels. Two ports are connected to a Panasonic PBX (it's working fine), and others two ports are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected to a CO port (where i had a pstn line). When I call to the Alcatel PBX, the asterisk show me in it console that es ringing but

All the broadcasters on the server which I support deliver their content in MP3 format. Recently, there's been interest in supplying a second AAC stream at half the bandwidth but with the same audio quality (64kbps AAC versus 128kbps MP3) like TuneInRadio does for delivering their content regardless of the source. I've thought of using a third-party product called Stream Transcoder, but am

With best regards Abdul Ahad Anwer Khan, M.Sc(CME, in progress) University of Applied Sciences Offenburg Germany Phone:+497814748226 Mobile:+4917623468462 From: abdulahadanwer at hotmail.com To: asterisk-users-bounces at lists.digium.com Subject: New in asterisk Date: Sun, 27 Sep 2009 14:50:59 +0600 Hello All I am a student and doing my thesis which is related to asterisk. I am new in

I've got a question about codec_dahdi witrh a system running Asterisk 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to route calls between different PRI connections, so no transcoding between codecs is happening as far as I am aware. 1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? 2) I'm

Hi, I would like to improve my understanding of T.38. 1. What T38FAX_VERSION_0 or T38FAX_VERSION_1 in chan_sip.c means ? voip-info.org implies one has to change values in chan_sip.c to make it work. Shall I set T38FAX_VERSION_1 or leave T38FAX_VERSION_0 in global_t38_capability ? Source code says "This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC,

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name

On 31 May 2003 at 12:49, Karl Heyes wrote: > On Sat, 2003-05-31 at 10:29, Stefan Neufeind wrote: > > I have mp3s as input and can't convert them all. And since my > > listeners will ONLY use mp3-streaming I don't want the overhead and > > quality-loss of converting mp3 to ogg and later stream-transcode > > them from ogg back to mp3. That's not worth it.