similar to: Javascript source client

Thomas, Thank you for your interest in this, you description is as accurate as I can see. > From my perspective your challenges will be to get the containers right. > WebM for audio+video > Ogg for audio > > Also (I'm not that familiar with webRTC) you might need to reencode > to Opus and VP8 in some cases? here is the great news

Hey all, So we have been advised from this thread https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 to not use http put as it is not in real-time, instead they are suggesting the use of SDP, is that something that icecast supports? Or does anyone have other ideas on this? ~stephen On Sun 12 May 2013 01:51:31 AM CDT, Thomas Ruecker wrote: > Hi, > > On 11

I'm following up on a thread started by Stephen a couple months ago about building a JavaScript source client using webrtc. The first step suggested was to figure out how to mux the audio and video. After I posted a feature request on the webrtc experiment js library, we seem to have a solution: https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-20791759 Based on the last

Hi, On 11 May 2013 04:44, Stephen Mahood <mv at cyberunions.org> wrote: > I am new to the dev list here, but my question specific about any > development towards webrtc integration. I'm currently not aware of any such development. > Let me explain, a couple colleagues and I are currently working on our > webrtc build at live.mayfirst.org site useing nodejs. We are

Hi, On 11 May 2013 15:32, Stephen Mahood <mv at cyberunions.org> wrote: > Thank you for your interest in this, you description is as accurate as I > can see. > >> From my perspective your challenges will be to get the containers right. >> WebM for audio+video >> Ogg for audio >> >> Also (I'm not that familiar with webRTC) you might need to reencode

Hi Stephen, > So we have been advised from this thread > https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 > to not use http put as it is not in real-time, instead they are > suggesting the use of SDP, is that something that icecast supports? Or > does anyone have other ideas on this? The imminent Airtime 2.4.0 release has support for Opus, and it

Five years after initially posting to this list[0], I finally completed as browser-based video source client iceast. The code is here: https://gitlab.com/jamie/icecream As Romain Beauxis responded to my initial email, webrtc was unsuitable and websockets was the way to go. Thanks for the help! jamie 0. http://lists.xiph.org/pipermail/icecast-dev/2013-July/002223.html -- May First/People Link

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

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Hey i have an interesting topic to discuss here. 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

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> 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? >

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

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

Begin forwarded message: > From: "Tom O'Reilly" <TOreilly at mpegla.com> > Date: May 22, 2010 6:31:50 PM CDT > To: "Dave Johnson" <davefilms.us at gmail.com> > Cc: "Info-web" <Info-web at mpegla.com> > Subject: RE: VP8 > > Dear Mr. Johnson, > > Thank you for writing. We appreciate hearing from you and the >

Hi all, Sorry for the VP8 question on the Theora list, but I think all the relevant people are here :) I'm using Icecast to distribute some Ogg Theora streams at the moment with a view to adding VP8 along side these in future. Ideally I'd just use Ogg VP8+Vorbis, I know there is a mapping for this already but how much support from the browsers can I expect for this configuration?

"Google will soon make its VP8 video codec open source, we?ve learned from multiple sources. The company is scheduled to officially announce the release at its Google I/O developers conference next month, a source with knowledge of the announcement said. And with that release, Mozilla ? maker of the Firefox browser ? and Google Chrome are expected to also announce support for HTML5 video

Patenting a mathematical formula is NOT creating a machine nor is it unique. For example. 2+2=4... apples + apples^2= given outcome. I want to patent this. It's stupid to patent something like that. The same is true for formula algorithms. Algorithms occur in nature. Thus should not be patented. Now, Volley G Mathison inventor of the Electropsychometer had a machine that he could patent. A

Hello , We are using Icecast for few years in a small french radio station with great success ! ( hitting 60 simultaneous listeners sometimes :-) ) http://www.radiogalere.org:8080/ Now we plan to stream the webcam capture of the studio, we 've done a test with Ogv/theora @128kb video with great sucess although none of the HTML5 browser wher able to keep on playing the stream after few

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

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 ~]#