similar to: Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

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, Has somebody get combination Asterisk (I'm using version 13.32.0, webrtc and iOS (version 13.4.1) with Safari or any other browser working properly in confbridge conference calls? I hope my Asterisk webrtc related settings are not totally wrong, because several other browsers from Windows seem to work. Best regards, Teijo

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

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:

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

hi, i have strange problem with asterisk 13.15.0+pjsip bundled/centos 7/systemd start script we are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days ago today i have problems with stopping/crashing asterisk /var/log/asterisk/messages dont show any clues [May 9 12:10:52] WARNING[25762] pjproject: tsx0x7fbb28024088

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

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

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?

Hi. I made a webrtc relay with recording and dumped the SDP requests and RTP packets into files. Then I made a java decoder based on jitsi. Although the files contain all the needed info: encription keys, codec info, timestamps, etc., I could only decode one side in one of 2 conversations. For example, the RTP payload is decrypted successfully, but opus_packet_get_nb_samples() or opus_decode()

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

On the call where even opus_decoder_get_nb_samples() fails, what's the value of the first two bytes of the buffer you're sending? And what's the length of the buffer? Jean-Marc On 11/29/2013 03:31 PM, Ilya Basin wrote: > Yes, I parse the RTP header, decrypt the payload and then feed the > decrypted data to opus. Besides, opus_decoder_get_nb_samples() on 1st > packet returns

Yes, I parse the RTP header, decrypt the payload and then feed the decrypted data to opus. Besides, opus_decoder_get_nb_samples() on 1st packet returns reasonable number 960, but then opus_decode() fails. On Fri, Nov 29, 2013 at 11:39 PM, Jean-Marc Valin <jmvalin at jmvalin.ca>wrote: > On 29/11/13 01:49 PM, Ilya Basin wrote: > > For example, the RTP payload is decrypted