similar to: How to use http-put for JavaScript source client

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

involved. This is easy enough to verify, dump to a file as described on that page and feed the resulting file to 'ogginfo'. If it is indeed properly muxed into an Ogg container, then just forwarding it via HTTP PUT instead of writing it to a file should do the job. I'd suggest someone just tries it. If you need a Icecast server to test against I can also provide that. Ping me on IRC

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

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

Hi, On 07/23/2013 07:44 PM, Jamie McClelland wrote: > 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: >

Thanks so much for the comprehensive response Romain. It sounds like you are working hard getting this working with audio streams and it's a lot more complicated than I had originally hoped. I'm still curious about the proper syntax of the http put command though (and I'm sure others would be interested as well). Is there any documentation on the parameters to send? I also had a

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

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

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

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

> What machine are you running (namely what OS)? Debian. > I dont understand your approach. > Why running a 'streamer' behind a nat? > Not enough 'resources' to rent/ rent to buy a ded. Server? > Mean, can't expect to satisfy a lot of listeners this way. :-) I am listening the Muazkhan indications XD: > > Hi Muaz Khan, >> We are adtlantida.tv and

Hey everyone, I am new to the dev list here, but my question specific about any development towards webrtc integration. Let me explain, a couple colleagues and I are currently working on our webrtc build at live.mayfirst.org site useing nodejs. We are currently looking into seeing if there is any development of a javascript source client? This would be used to send the webrtc room to a public

Hi list, The Savonet team is proud to announce a new release of its programmable audio stream generator, liquidsoap 0.3.0. Liquidsoap is a simple ruby-like script language allowing one to build audio stream sources from various elementary sources, source combinators and audio outputs. It is mainly intended to be used as an icecast client for internet radios through the use of the shout output

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

In the asterisk wiki instructions for Configuring Asterisk for WebRTC clients... https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients "To communicate with websocket clients, Asterisk uses its built-in HTTP daemon. Configure /etc/asterisk/http.conf as follows: [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089

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, On Sun, Sep 7, 2008 at 12:30 AM, Dick Trump <dtrump1 at triadav.com> wrote: > kosnickx wrote: >> But there is one thing : while falling back from one mount point to >> another it takes about 10 to 15 seconds. Does this happen to you too? Is >> there any way to reduce this to minimum or to 0 if possible. I've been wanting to advise you to give liquidsoap a try

I'm learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side. I noticed that the Asterisk wiki mentions the need for res_pjsip_transport_websocket, so does that mean Asterisk requires the signaling to occur over a websocket? If I used

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

Hi list, I'm the main developper of liquidsoap, the streamer from the Savonet project (http://savonet.sf.net). This streamer mainly uses shout2 as an output plugin, but also alsa, raw data over rtp, etc. And many outputs can be used in the same instance of liquidsoap. For that reason, we don't use directly sync and delay from libshout, but have a centralized scheduler, which should avoid