search for: trancoding

Displaying 14 results from an estimated 14 matches for "trancoding".

Did you mean: transcoding
2004 Aug 26
4
Codec
Good day all I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem Pleas help ALtus
2009 Feb 26
1
codec_dahdi and Asterisk 1.6.0.6
...ahdi channel to another dahdi channel? 2) I'm getting the following error on the Asterisk CLI unless I disable the codec_dahdi module: ERROR[18854]: codec_dahdi.c:399 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory Is this because I do not have a hardware trancoding device? Can I safely ignore this error or is it a bug? Brandon. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090225/11fb3dfe/attachment.htm
2016 Apr 08
1
Icecast and AAC streams
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
2016 Mar 04
2
Icecast and AAC streams
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
2004 Aug 06
2
no mp3s with ices2!? an other way?
...hat's not worth it. > So why can't support for mp3 be added to ices2? From a technical > point, I mean? Somebody said "It's not and never will be". Any good All transcoding suffers from this. it doesn't matter if you do it before hand converting all your files or trancoding on-the-fly inside something like ices. Nothing is stopping it technically, but it's the non-technical points that are stopping this, there are patents on mp3 that people need to consider, ices2 is a xiph project so producing ogg streams comes before sorting out mp3 and of course some developer...
2004 Aug 06
0
no mp3s with ices2!? an other way?
...> > So why can't support for mp3 be added to ices2? From a technical > > point, I mean? Somebody said "It's not and never will be". Any good > > All transcoding suffers from this. it doesn't matter if you do it > before hand converting all your files or trancoding on-the-fly inside > something like ices. > > Nothing is stopping it technically, but it's the non-technical points > that are stopping this, there are patents on mp3 that people need to > consider, ices2 is a xiph project so producing ogg streams comes > before sorting out mp...
2014 Feb 03
0
Relay/forward RTP-packets over icecast2
> What machine are you running (namely what OS)? Debian. > I dont understand your approach. > Why running a 'streamer' behind a nat? > Not enough 'resources' to rent/ rent to buy a ded. Server? > Mean, can't expect to satisfy a lot of listeners this way. :-) I am listening the Muazkhan indications XD: > > Hi Muaz Khan, >> We are adtlantida.tv and
2016 Mar 04
0
Icecast and AAC streams
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.
2009 Feb 09
1
What t38pt_udptl is ? Explain T.38 in 1.4
...anding 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, Version 0 and 9600 max fax rate" What does "Version 0" relates to in this aboxe line ? MMR and JBIG transcoding ? 2. A softphone like Zoiper is able to receive voice or fax calls. When setting those values (see bellow) in sip.conf,...
2014 May 27
0
dahdi-dahdi native bridging and audio level
...4 Initially I had no txgain=5.0, but users complains that they hear not as good as before asterisk was added into this scheme. We even tested and found that yes, asterisk decreases volume. Could you tell me why? As I understand this is digital connection and there is the same alaw codec, so no trancoding. Why can be sound volume changed during chan_dahdi native bridging? Thank you!
2005 Jun 07
2
codec preference
Need some help understanding codec preferences: I have 2 asterisk servers. Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and allow=ulaw in iax.conf Server 2 receives calls and routes them to server 1. It has the same allow lines. We receive calls from a phone co and route them via server 2 to server 1. The calls originate in g729 and everything works fine. Now I want to take
2005 Jun 03
1
oh-323 / Cisco AS5300 problem
Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and
2004 Aug 06
4
no mp3s with ices2!? an other way?
> Best wishes > Stefan (on second thought ) :: my last post assumed that you had .OGG files to begin with .. depending on the size of your library, you may want to convert your mp3s to ogg . i ended up doing this , and it made things much simpler on the web end . also ; check out dir2ogg .. echo "dir2ogg written by Darren Kirby; linuxbot@shaw.ca; www.badcomputer.no-ip.com/"
2019 Jul 05
4
Asterisk and Linphone
Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the