search for: transco

Displaying 8 results from an estimated 8 matches for "transco".

2009 May 04
1
virtual mailbox users users can send, but can't read e-mail
Greetings, I am trying to add the second virtual mailbox domain (transco.org.au) to an existing Postfix/Dovecot/MySQL mail server. Users of the first virtual mailbox domain (transylvania.org.au) have no problems sending and receiving mail. Users belonging to the second virtual mailbox domain can successfully send mail to outside, however they won't receive the...
2013 May 31
3
Opus vs. Vorbis
Hi all, I've been looking at opus lately as a replacement to vorbis and it sure seems to me that, at least on paper, it is the 'next generation' streaming codec. Has anyone been using opus as a streaming codec to Icecast? If so, how do the following general characteristics compare to vorbis? + Quality + Latency + Bandwidth + Performance (on stream client and Icecast server) +
2013 May 31
2
Opus vs. Vorbis
...good. > >Latency is lower, but for broadcasting a radio program for me at the >moment it is not that importend, so i did not test that really with >icecast (i did test opus low latency but with other applications: >fideliphone and teamtalk) > >Stream runs stable, my stream is a transcode made by Liquidsoap, from a >transparent ogg vorbis stream (The reason i'm doing that is because i'm >transcoing also to mp3, vorbis aacplus at different bitrates). > >Input is a live stream from a studio to my main icecast server, on this >server i transcode with liquidsoa...
2013 May 31
0
Opus vs. Vorbis
...r bitrates the codec sounds good. Latency is lower, but for broadcasting a radio program for me at the moment it is not that importend, so i did not test that really with icecast (i did test opus low latency but with other applications: fideliphone and teamtalk) Stream runs stable, my stream is a transcode made by Liquidsoap, from a transparent ogg vorbis stream (The reason i'm doing that is because i'm transcoing also to mp3, vorbis aacplus at different bitrates). Input is a live stream from a studio to my main icecast server, on this server i transcode with liquidsoap. Then every stream...
2003 Apr 14
3
Progress Bar
.... Is this a deliberate change to "scp"? John Durkin ______________________________________________________________________ Unless expressly stated to the contrary, the views expressed in this email are not necessarily the views of National Grid Transco plc or any of its subsidiaries or affiliates (Group Companies), and the Group Companies, their directors, officers and employees make no representation and accept no liability for its accuracy or completeness. This e-mail, and any attachments are str...
2013 Jun 09
0
Opus vs. Vorbis
...cy is lower, but for broadcasting a radio program for me at the > moment it is not that importend, so i did not test that really with > icecast (i did test opus low latency but with other applications: > fideliphone and teamtalk) > > Stream runs stable, my stream is a transcode made by Liquidsoap, from a > transparent ogg vorbis stream (The reason i'm doing that is because i'm > transcoing also to mp3, vorbis aacplus at different bitrates). > > Input is a live stream from a studio to my main icecast server, on this > server i tran...
2013 May 31
1
Opus vs. Vorbis
...> > Latency is lower, but for broadcasting a radio program for me at the > moment it is not that importend, so i did not test that really with > icecast (i did test opus low latency but with other applications: > fideliphone and teamtalk) > > Stream runs stable, my stream is a transcode made by Liquidsoap, from a > transparent ogg vorbis stream (The reason i'm doing that is because i'm > transcoing also to mp3, vorbis aacplus at different bitrates). > > Input is a live stream from a studio to my main icecast server, on this > server i transcode with liqui...
2004 Apr 12
0
RE: Asterisk-Users digest, Vol 1 #3408 - 12 msgs
...gt; > > No. Once you pay the G723, patent holders their money I'm pretty sure the unoptimized ITU sample code for the G723.1 codec can be added to Asterisk pretty easily. But since it's not optimized don't expect to be able to send many G723.1 calls thru Asterisk if you are transcoding. Of course, if you are NOT transcoing, then this whole thread is moot since Asterisk already supports G723.1 passtthru. See this message in the mailing list archives: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001317.html You can also pay the $30 or so to the ITU to get a...