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9

I've been a systems administrator for a TV+radio station a few years ago, and I also had lots and lots of analogue broadcasting units in my server room. What I did was to make sure that every single piece of broadcasting equipment was properly grounded. I got an electrician to run a new, isolated ground feed to the server room, which I then terminated at ...


6

Your setup is suboptimal -- I'd even go so far as to say that the setup you described is actively hostile to VoIP communications. A few key items: 1. You're sending video and audio both ways over the RDP session. 2. You're going over a wireless network. While this often works fine it's generally not a great idea, especially on common WiFi (not dedicated ...


5

Audio is special in a lot of ways, and audio drivers run in the context of a "session" in Windows, so each remote desktop user gets their own audio. Even on a physical machine, when you connect via remote desktop, you see different audio devices in device manager from the ones you see sitting a local keyboard. The short answer to your question is that a ...


5

It does seem like it is EMI. Your use case is quite unusual so it's unlikely many people will have encountered it. As you have determined that it is the new server that is causing the problem by the simple expedient of turning it of and on then (I'm guessing you have no support) you're going to have to diagnose the hardware yourself. If you have other ...


4

1) VNC does not transmit audio. 2) I'm going to assume that you're trying to open RDC and VNC sessions to the same computer at the same time. I'm also going to assume that it's an RDC session to the "console" rather than to a terminal server. When you connect via VNC, the OS sees that as the same thing as someone using the connected keyboard and mouse. ...


4

It's not ejabberd what handles audio/video talk. It's handled with Jingle (XEP-0166), which is client-to-client protocol extension for XMPP. The clients have to support it.


4

Red Hat doesn't distribute MP3-enabled packages due to patent and licensing issues. You can find the necessary packages for CentOS and Fedora from the RPM Fusion free and nonfree repositories.


3

It is not possible to transmit audio from a client computer to a Windows 2003 Terminal server. It is not supported. I suppose there is a remote possibility that you could get Skype to accept a call and receive audio, but there is no possibility for you to transmit. Windows 2008 R2 introduced a feature, to permit audio redirection, but it is not enabled on ...


3

I believe that FFserver can do what you're looking for. It's a part of the open source cross-platform FFmpeg suite, available at http://www.ffmpeg.org/


3

I've used many versions of asterisk, with several different VoIP providers (SIP and one or two IAX2), physical cards on analog lines (Sangoma/Digium/Pika), and many different PC-based and hardphones (Aastra/Polycom/Sipura/Linksys/Snom/Mitel) and have never had that problem. It's almost certainly your PSTN gateway. If it's an analog card, there's probably ...


3

try setterm -blength 0 man setterm


2

I used to use Jinzora which handles transcoding and all kinds of audio formats and has similar operation to Ampache. It has been a while since I used it, and I never did transcoding but it might be worth checking out.


2

There is a small tool called pcaputil - it is part of the pjsip project. It should be able to decode pcap files with RTP (G.711, G.722, speex and other codecs are supported) into wav files. Compile pjsip and find pcaputil inside pjsip-apps/bin/samples/[architecture]/.


2

Check out qtplay. Edit: Oops, I forgot about the built-in afplay(1). The man page tells you less than the usage statement, though. $ afplay -h Usage: afplay [option...] audio_file Options: (may appear before or after arguments) {-v | --volume} VOLUME set the volume for playback of the file {-h | --help} print help { --leaks} run leaks ...


2

A few thoughts; What's your client software, this is very important, specific versions on specific OSs - this will define your streaming software. What are your stream profiles, do you intend to pre-encode or do you have/expect-to cross-encode on the fly? Have you done your maths on average and burst bandwidth requirements for this, is your top end figure ...


2

Try VLC: http://www.videolan.org/vlc/ The most important advantages are: CLI interface + GUI, nearly all OS, remote control over web interface


2

My thoughts on this would be a problem between the codec, and the driver for the sound card. Unfortuantely windows XP 64bit is renowed for bad driver support. I would suspect that that particular stream uses a codec thats different to anything else you use, for streams that are working, and its having a problem when working with your sound card driver. Is ...


2

You can try with ffmpeg: ffmpeg -i file.wav file.mp3


2

I use Ericom Blaze as a low-latency RDP accelerator solution for my personal systems. I've also deployed it in situations where I needed full-screen or multiple-monitor RDP to a remote site. One clear example was having financial traders in Chicago who needed to run a GUI app in a co-located system in Toronto. Ericom's protocol was very efficient for this ...


2

Take a look at this link, it will help you. Pay attention to the last column. http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml However g729 is not royalty free from what I understand. I use g711, the quality is great and at 64k per call you would need 8 calls to fill your pipe. g729 is most effective and utilizes ...


2

upload your video via sftp, ftp or ssh. Move the video on a place with access/read permissions on the server... if your main site on the www.mydomain.com is /var/www/wordpress/ put the file on this directory. The URL is like www.mydomain.com/video.mp4. all depends of the directory permission on your site (ngix configuration).


2

I tried the files on my nginx server and all of them except flac work on android 4.4 with the default Browser and Firefox. This is the tag I used: <audio preload="auto" controls=""> <source src="Test/Audio/mp3.mp3" type="audio/mpeg"> </audio> Maybe you have not set your mime types in the nginx configuration correctly. You will need: ...


1

AirFoil would be my suggestion - http://www.rogueamoeba.com/airfoil/ It's basically the same idea as the Apple Airport express, but in software. One computer 'serves' the audio, all the others are clients.


1

Icecast most certainly supports MP3. Actually, the SHOUTcast-compatible servers do not care what the data type is. You can send them anything. (Don't believe me? Telnet into one, send some text, and watch it come out the client side!) Your viable choices: SHOUTcast DNAS 2.0 Icecast If you have problems with either of those, perhaps you should post ...


1

the id3v2 program is a powerful command-line tool to edit id3 tags. MusicBrainz picard is a GUI tool that can also lookup a central mp3 signature repository.


1

A combination of Digital Rapids various products would do what you want but not for free, not sure of any open source equivalents.


1

Check out liquidsoap. It can schedule multiple input streams and create file-based streams from an mp3 playlist. You could use a simple flash player like ffmp3 A flash-bashed admin interface is not available for liquidsoap.


1

The problem seems to have been that I was running this on a 256 slice. I've since moved to RackSpace Cloud Servers and am running an identical setup on a 4GB virtual server and I have no audio issues. Guess Asterisk needs a bit more muscle behind it.


1

There are two versions of Virtual Audio Cable. Version 4 does not support RDP but version 3 supposively does. Have you tried version 3? http://software.muzychenko.net/eng/vac3.htm


1

Do the servers have sound cards? If not, then that's the problem. You need to install a sound card and drivers before you can get audio capabilities.



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