similar to: Webrtc and iOS devices

Hello, Currently audio conference. Should upgrading Asterisk from 13 to newer version resolve webrtc/iOS problem? Best regards, Teijo Dan Jenkins kirjoitti 28.4.2020 klo 12.18: > First things first, upgrade from 13 - WebRTC has moved a long a lot since > then. If you can't upgrade everything to 13 then run another asterisk > specifically for WebRTC and bridge to your other

Hello, I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only problem until now which remained was that if dtls_rekey was set to the value other than 0, call hanged up when using chrome after the time where dtls_rekey was set. I suppose that "bad media description" shown in Chrome's window which causes call to fail, has appeared with Chromes newer versions

I honestly couldn't tell you if it would resolve it but there aren't many people going to be willing to help problem solve anything if you're running 13 - you'll get more support on 17 for example. Very easy to bring up a new instance or VM in the grand scheme of things to test the theory and get it working on most recent version of Asterisk On Tue, Apr 28, 2020 at 11:37 AM

First things first, upgrade from 13 - WebRTC has moved a long a lot since then. If you can't upgrade everything to 13 then run another asterisk specifically for WebRTC and bridge to your other Asterisk Is this just an audio conference? On Sun, Apr 26, 2020 at 10:21 PM Teijo <g.aloitus at gmail.com> wrote: > Hello, > > > Has somebody get combination Asterisk (I'm

The Asterisk Development Team has announced the release of Asterisk 13.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.10.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

Hello, Webrtc stopped after upgrading firefox from version 36 to version37. I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 and firefox version 36 without any issues until firefox was upgraded to version 37. Unfortunately Chrome works well in one direction (from chrome to any extension) but calling from an extension to a webrtc on chrome has one way voice. Could someone try

Hello Opus community, I'd like to ask you for advice and recommendations. WebRTC uses Opus, and I noticed https://webrtc-codereview.appspot.com/5549004 started referring to currently internal Opus headers. This is possible because for Chromium the Opus sources are just checked in, so any header can be #included. I detected this when trying to package Chromium for Linux distributions with

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct

I work on the Asterisk side of things and admit to not knowing about browser development. A co-worker asked me today why they should develop a web based agent software using WebRTC? They prefer to develop using a SIP based javascript library they found. Can anyone offer some insight on why to choose either WebRTC or a SIP library for a web based agent software connecting up to an asterisk

Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml

A while back (last year maybe?), there was a Digium blog post on setting up WebRTC. I was never able to get that working. I was working with Asterisk 15 on a RHEL derived distro and had no idea of where to go to shoot the failure. Has anyone got a tutorial with trouble shooting?

I am having a small problem that is driving me nuts. I can make calls over my Webrtc client without any problems and audio sounds fine. The only problem I have is that when I call an internal SIP extension on my PBX I do not hear the ring while I wait for the call to be answered. My dial command does include the rR options. If I make an external call to a land line or a mobile phone I do

hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from

2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo page on my own private >> Asterisk server and access this demo

Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.<div><br></div><div>I configure my Asterisk 11.7.0 to work wit WEBRTC.</div><div><br></div><div>Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards