search for: transcoing

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 mail, though Postfix

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) +

...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 is relayed by two >public icecast servers where listener clients connect to. > >Running this for q...

...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 is relayed by two public icecast servers where listener clients connect to. Running this for quite a while now and it...

Product: Portable OpenSSH Version: 3.6p1 and 3.6.1p1 Platform: ix86 OS/Version:?Solaris 8 Problem: When copying files between networked systems using "scp", no asterisk characters are displayed on the progress bar as in previous versions of OpenSSH. Is this a deliberate change to "scp"? John Durkin ______________________________________________________________________

...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 is relayed by two > public ic > ecast > servers where listener clients conn...

...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 is relayed by two > public icecast servers where listener clients connect to. > > Running this f...

...3, 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 copy of the G723.1 specification and sample...