similar to: Chrome 33 released with Opus support in HTML <audio>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 In addition to Chrome's support for Opus in WebRTC, Tom Flanagan has added support for .opus files in the HTML <audio> element, similar to what Firefox has. It's currently behind a switch, so you'll need to pass --enable-opus-playback to M25 or M26 to evaluate. I tried some the files https://people.xiph.org/~greg/opus_testvectors/

hello all - sorry for such a basic newbie question, but is opus now fully supported using the latest google-chrome? according to this link: http://news.cnet.com/8301-1023_3-57577464-93/google-hitches-opus-audio-technology-to-webrtc-star/ "Chrome 27, making its way through the development pipeline, is helping to advance the fortunes of a?new audio compression technology called Opus."

I'm having the same issue! The difference in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00

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 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 ~]#

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

Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > Hi Oliver, > > On 02/18/2016 12:10 PM, Olivier wrote: > > 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

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()

Toufic Khreish (Gmail) wrote: > 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

Hi, I work on a webRTC application and recently tried updating from opus 1.1.5 to 1.2.1 Afterwards I noticed occasionally weird audio glitches. I finally tracked down the issue to the opus decoder in my application outputting samples with the value of -32768. This behaviour stopped when reverting to opus 1.1.5 or compiling opus 1.2.1 without configuring --enable-float-aprox and --0fast. The

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

Any luck reproducing the problem with opus_demo or opus-tools? Jean-Marc On 02/22/2018 10:14 PM, Stepan Salenikovich wrote: > > > On Thu, Feb 22, 2018 at 9:53 PM, Jean-Marc Valin <jmvalin at jmvalin.ca > <mailto:jmvalin at jmvalin.ca>> wrote: > > On 02/22/2018 09:34 PM, Stepan Salenikovich wrote: > > Its unexpected because the decoder continues to

Hello, I am trying to encode an Opus file with Ambisonics including Head-Locked (non-diegetic) Stereo sound for a Virtual Reality 360° video. YouTube describes the spatial audio requirements here: https://support.google.com/youtube/answer/6395969 It's the last list item 5. > 5. Supported First Order Ambisonics (FOA) with Head-Locked Stereo format: > W, Y, Z, X, L, R as a 6-channel

Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the

Hi all ! I am using sipML live demo page (http://sipml5.org/call.htm?svn=224#) in order to test WebRTC setup on my Asterisk PBX. I am using latest SVN version of Asterisk 11 (Asterisk PBX SVN-branch-11-r417677) If I make calls from softphones (Zoiper, X-Lite), which do not support DTLS at all, I can hear the Echo Test sound. BUT when I call from browser (I've tried latest Mozilla Firefox

On Thu, Feb 22, 2018 at 9:53 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 02/22/2018 09:34 PM, Stepan Salenikovich wrote: > > Its unexpected because the decoder continues to output all samples > > of -32768 even when the microphone input is silence or near silence, so > > I would expect the decoded values to be at or near 0. > > Oh, if the output is

On Thu, Feb 22, 2018 at 8:34 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Hi Stepan, > > I would need more information to be able to investigate further. It's > legal for the decoder to output -32768, so it would be good if you could > explain how this is unexpected. Its unexpected because the decoder continues to output all samples of -32768 even when the

Following up on this topic ( sorry if this starts a new thread but I just joined the ml ), I do no understand why it is not possible to use the audio stream from webRTC's getUserMedia and then send it over a websocket ? It seems that the webRTC implementation can natively encode in ogg format in stereo from any interface ( according to

Hi Jamie, The webRTC API does not sound suitable for source->server streaming for many reason. For instance, the peer-to-peer connection requires input from both end and seems quite unfeasible to implement in a server. Likewise, codecs are completely abstracted and much more. In reality, webRTC is an API to acheive full-duplex conversations a-la skype and not for streaming. For these

The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine