similar to: g.722 + loudness

Hi, G.722 is heavily used by Broadcasters worldwide for wideband voice communications over ISDN. I'd like to be able to receive these G.722 over ISDN calls into an Asterisk exchange (with mostly a view to routing the calls to a Voicemail box where material can be recorded). I have been examining source code for the 3 different ISDN Channels in Asterisk and they all seem to be hard- codec

Hi all, I am having problem with conference call (meetme feature) using G.722 phone. G.722 phone to phone is working fine. I suspect this is due to the fact that Asterisk 1.4 only support G.722 passthrough. Any ideas how this problem can be fixed. Thanks. Regards, Chong -------------- next part -------------- An HTML attachment was scrubbed... URL:

For about 10 months I have been running a succession of Asterisk SVN trunk versions on an Athlon 64 X2 4400+ based machine with OpenSuSE 10.2 at my home. I have a variety of SIP phones (mostly Polycom) internally; my external connections are two POTS lines on a TDM400P (with HPEC) and an IAX2 link to a VoIP provider. I had Asterisk configured to allow G.722 and u-law on the Polycom phones,

For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel

The G.729.1 "wideband" codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it

Situation:? * remote virtual server with very little storage (estimate: I can spare about 40G for music) * local music collection of ~80G in all sorts of formats - lossy in varying quality, some lossless too Vision: * stream my 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?

Opus or AAC will give you comparable results at reasonable bitrates (~128k). Though, I would suggest finding a way to get more storage. You could upload to Backblaze B2 or AWS S3 for pennies, if your current host won't let you upgrade. On Sat, Apr 15, 2023 at 3:36?PM D.T. <ohnonot-github at posteo.de> wrote: > Situation: > > - remote virtual server with very little

I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is high. It produces loud clicks as if clipping. For quiet audio however, it seems fine. ADPCM (Digilogic VOX?) seems to be

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac

Been playing with G.722 in Asterisk 1.4.24.1 - using the back-ported patches from http://carlton.oriley.net/drupal/node/12 Works just fine as far as I can tell - Grandstream phones anyway - playing the G722 sound files, and calls between them. Transcoding seems fine too - calling non G722 devices, it seems to "just work" However Phone A (g722) calls phone B (gsm). Works fine.

Hey all, I have followed a thread on golem.de, which was about an article regarding mozillas reasons, not to include h264 and to prefere theora instead. In the forum there was much talking about a lot of nonsens (as usual). But there is still a huge and loud number of people believing that theora has a significant worse quality compared to h264. Most test material I found does not focus real

I'm trying to figure out which 1.6 releases have the fix I'm looking for by reading the ChangeLog for each release, whether it's in the 1.6.0, 1.6.1 branches, or an -rc release. If I look at the latest -rc releases of 1.6.0 and 1.6.1 (which are 1.6.0.11-rc2 and 1.6.1.3-rc1 respectively), will that be an exhaustive list of changes or not? The reason being I'm still waiting on the

Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for

It appears that both the echo canceller and the noise reducer (NR) introduce rather severe clicking artifacts when presented with audio that has slightly overloaded the A/D converters. I am talking about speech that sounds just slightly distorted, due to clipping, when simply played back. If I pass that speech through the echo canceller or the noise reducer, it acquires really loud clicks and/or

Can anyone recommend some kind of virtual t.38 fax software? I'd like to test/debug some of the t.38 stuff, but it'd be much easier if I had a software client that could just generate the faxes from a workstation, rather than having to sit with the fax machine + t.38 ata to source faxes from. There doesn't seem to be much out there, and the stuff that's out there is kind of

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

Working on a paging system for one of my sites and running into something I can't believe is this hard. In one of the zones, they want to have three different extensions ring over the pa system, using it as a loud ringer. Now the paging system does have a loud ringer built in and I can easily have it do a simultaneous ring, but all of the extensions will sound the same over the loud

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What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate

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 transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace.