similar to: Integrating 1.1-beta

Hi All, As I've said before, Opus is excellent. I use it to reduce network load when running a remote radio, the encoding improvements in 1.1 are very impressive. I downloaded and compiled 1.1 (VS2010) but can't find the 1.1 API changes - maybe I'm just being extra-thick this morning J . Simon GD4ELI/HB9DRV http://v2.sdr-radio.com/ -------------- next part

Boris, 1.1 beta is working great on Windows in my client / server SDR radio software. Testers say encoding is much better than 1.0 (which was good anyway). Simon GD4ELI/HB9DRV http://v2.sdr-radio.com/ -----Original Message----- From: opus-bounces at xiph.org [mailto:opus-bounces at xiph.org] On Behalf Of Boris Grozev We've been using Opus for VoIP calls in Jitsi and we're very happy

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, We just released Opus 1.1-beta and Monty wrote a nice demo showing off all the new features, including many improvements over the alpha release: http://people.xiph.org/~xiphmont/demo/opus/demo3.shtml As usual, the code can be downloaded from: http://opus-codec.org/downloads/ In addition to 1.1-beta, we also released 1.0.3, which includes a

Hi, a first set of results can be downloaded here <http://trac.tools.ietf.org/wg/codec/trac/wiki/TestingResults> http://trac.tools.ietf.org/wg/codec/trac/wiki/TestingResults More will be added soon. Thanks to Patrick Schreiner and Alfons Martin, who did the tests. With best regards, Christian Hoene Symonics GmbH Sand 13 72076 T?bingen Tel +49 7071 5681302

On 18/07/13 04:13 AM, Boris Grozev wrote: > We are thinking of updating to 1.1-beta and wanted to know if there's > any reason not to do so at this point, and wait for the 1.1 release > instead. Is the beta expected to be as stable as the 1.0.x series, or > are there any known flaws still to be corrected? There's no *known* issues with 1.1-beta compared to 1.0.x and we do

Hi All, I'm wondering about saving audio to a file. With MP3 & WMA there's support for tags - is there a file format for Opus with tag support? I guess I've missed something somewhere. Opus is excellent BTW, I use it in client/server radio communications software, http://v2.sdr-radio.com/ . Simon G4ELI/HB9DRV http://dit-dit-dit.com -------------- next part

Le 03/05/2020 ? 08:44, Benson Muite a ?crit?: > They have rpms: > > https://download.jitsi.org/jitsi/rpm/ That's the Jitsi desktop application. > > and also a scalable installation: > > https://github.com/jitsi/jitsi-meet/blob/master/doc/scalable-installation.md That's Debian-specific. > > Which one do you need? None. I need a comprehensive

Benson, no SELINUX was not enabled. The instance was selected without it just to make things easier. I do not have a pull request for the installation manual yet. On Tue, May 5, 2020 at 1:21 AM Benson Muite <benson_muite at emailplus.org> wrote: > > On Mon, May 4, 2020, at 10:38 PM, Erick Perez - Quadrian Enterprises wrote: > > Hi Centos friends. > > I had some time to

On May 3, 2020 3:33:29 AM EDT, Jason Pyeron <jpyeron at pdinc.us> wrote: >> -----Original Message----- >> From: Behalf Of Nicolas Kovacs >> Sent: Sunday, May 3, 2020 2:54 AM >> >> Le 03/05/2020 ? 08:44, Benson Muite a ?crit : >> > They have rpms: >> > >> > https://download.jitsi.org/jitsi/rpm/ >> >> That's the Jitsi

Hi Centos friends. I had some time to write a spartan tutorial on running the latest stable Jitsi Video Bridge and Jitsi Meet and Centos 7.7. I wrote it while testing it so this WORKS and I am currently using it for fun with the kids. I do have the server currently running but blocked by my firewall. I am willing to allow a few of the people such a Kovacs and others to connect to my Jitsi server

Hello, 1. I'm looking for an (anomized) example of a Jitsi Desktop provisioning file compliant with Asterisk ? Jitsi Doc mentions it should adhere to Java properties file syntax (see [1]) but a working example would help. If this example file included the following settings, it would be perfect: - sip username, secret, authname, ... - codec G711, RFC2833 - BLF - local or remote phonebook -

Hi, Anyone here managed to run Jitsi Meet on a CentOS 7 server ? As far as I can tell, it's one of the nicer open source videoconferencing solutions out there. I gave it a spin on https://suricate.tv and now I'd like to install it on one of my servers. Unfortunately the Jitsi developers only provide documentation for Debian/Ubuntu. Well, there is some sort of guide for CentOS around. As

On Sun, May 3, 2020, at 10:33 AM, Jason Pyeron wrote: > > -----Original Message----- > > From: Behalf Of Nicolas Kovacs > > Sent: Sunday, May 3, 2020 2:54 AM > > > > Le 03/05/2020 ? 08:44, Benson Muite a ?crit : > > > They have rpms: > > > > > > https://download.jitsi.org/jitsi/rpm/ > > > > That's the Jitsi desktop

I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the "-4" command line option to use IPv4 only just in

I have been challenged by someone that opusfile's 48kHz most of the time forced resampling but I didn't have good arguments to counter. I've checked the opus-codec.org pages and Monty's pages without finding good arguments. If you encode files to .opus with opusenc and decode with opusfile's api the sampling rate is constantly 48kHz ignoring output hardware isn't it? Is

Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call

I am thinking of buying a Logitech C920S Pro HD or C922 webcam for the necessary video conferencing. Is anyone using that with Zoom or perhaps with Jitsi? Do I need drivers? Any issues? Thank you.

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536

Hello, We have a use case where we want to record an opus RTP stream to a .opus file. We want to fill in any gaps in the stream, and we also want to take advantage of inband FEC whenever possible. The ogg/opus draft describes[1] how to fill in gaps by generating zero-byte frames, but I do not understand how (and if) FEC can be used. Is this possible, and if so, what is the recommended way of