search for: transcoding

In case there are any Debian developers around here, I wanted to point out message <20020828154322.GA15114@chulak.naquadah.org> on the debian-devel list this morning. Another Debian developer is proposing to submit an mp3->vorbis transcoding program for inclusion in Debian. I have objected to this on the grounds that the resulting vorbis files will sound like crap, and I have also pointed out that the vorbis developers have a low opinion of this sort of transcoding (I think this is an accurate representation of the developers' view...

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963...

...reat for inter-* connections when bandwidth is an issues (from what I read). G.711 finally is required since that is the only common protocol between * and the IP phone available. But then I stumbled across the passage I quoted below. Should I reconsider the setup to at least remove one of the transcodings? Or is the document's author simply wrong? Greetings, Philipp 2) Transcoding: To be avoided at all times Transcoding is the conversion of a voice stream with one codec to a voice stream with another codec (e.g. G.729 to G.7.23). Transcoding dramatically degrades the voice quality. It ha...

I have a CentOS release 4.7 box running asterisk-1.4.26.1 with dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0 I regularly get these messages, is this something i should be worried about? [Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Aug 21 01:05:07] ERROR[4343] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory [Aug

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

...g729 channel, the audio is complete rubbish, with a lot of stutters in it (sounds like a recording does when you upload a file in the wrong sample rate etc) - the file that we are playing back is a wav file that has existed on the system and has been successfully played back with the soft g729 transcoding and also plays back fine when the channel is alaw, just not when the channel is g729. The same issue occurs when a transcode has to happen from a handset to a IP trunk, eg alaw on the handset and g729 on the trunk channel, the audio stream is non comprehensible. The other issue is that whilst...

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CP...

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g72...

Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in the filename, does MixMonitor transcode both legs of the stream to a format it can then "mix" then transcode it back to g729 to write to disk? Thanks...

I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal

Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 ..... and forward it to a media gateway .. Regards Khaled chehab ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw /var/lib/asterisk/sounds/fr/service_notactivated...

As far as I can see, transcoding could be usefull for people who do not primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcod...

...whole music collection randomized so I can listen to it anywhere Plan/Idea: * Locally transcode everything to one format that results in files that are? - small enough to fit on my server, altogether - have a reduced bitrate for streaming - can be streamed as-is without further transcoding * Upload * Set icecast up to do just that (this I know how to do) So I'm asking advice for the transcoding. What's likely to give the best results with already lossy sources, and at small bitrates? According to these documents: https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio https:...

Dear ALl Can we use Asterisk for only for transcoding?. if yes how many concurent call we can transcode with help of Astetrisk? For this we only need to config SIP.conf or any other file too. Thanks Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/201...

Hello I use asterisk realtime, and I can set the order of codec preference on my realtime allow column. If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on CPU) - and more or less, force the codec to use by setting the codec preference So, can I disable transcoding?

...gt; anywhere > > > Plan/Idea: > > - Locally transcode everything to *one* format that results in files > that are > - small enough to fit on my server, altogether > - have a reduced bitrate for streaming > - can be streamed as-is without further transcoding > - Upload > - Set icecast up to do just that (this I know how to do) > > > So I'm asking advice for the transcoding. What's likely to give the best > results with already lossy sources, and at small bitrates? > > According to these documents: > https://trac...

Asterisk Project Security Advisory - AST-2019-005 Product Asterisk Summary Remote Crash Vulnerability in audio transcoding Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Minor Exploits Known No...

...> anywhere > > > > Plan/Idea: > > * Locally transcode everything to one format that results in files > > that are? > > - small enough to fit on my server, altogether > > - have a reduced bitrate for streaming > > - can be streamed as-is without further transcoding > > * Upload > > * Set icecast up to do just that (this I know how to do) > > > > So I'm asking advice for the transcoding. What's likely to give the > > best results with already lossy sources, and at small bitrates? > > > > According to these do...

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 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...