Rtcpeerconnection jobs

Hello, I have a laravel project using opentok to use webrtc. I get this error on Chrome: Cannot acquire video: :: NewPeerConnection: Publisher PeerConnection with connection (not found) failed: Failed to create PeerConnection, exception: NotSupportedError: Failed to construct 'RTCPeerConnection': Plan B SDP semantics is a legacy version of the Session Description Protocol that has severe compatibility issues on modern browsers and is no longer supported. See for more details, including the possibility of registering for a Deprecation Trial in order to extend the Plan B deprecation deadline for a limited amount of time.. Please check platform requirements and reload the page, or contact support. I need to fix this today

Requirements This app uses the following Javascript libraries: jQuery Bootstrap This app uses the following PHP library: Ratchet In order to run Magnoliyan Video Chat PRO server side you need: https (ssl certificate). Since Chrome 47 getting a camera/mic access is allowed only over https. The s...order to run Magnoliyan Video Chat PRO client side you need to: host it on a domain (even local/private virtual domain can work for testing purposes) for facebook authorization, create your own facebook login app pointing to that domain. Please read this tutorial for more information newer modern browsers like Chrome or Firefox (excluding IE) which support HTML5 technologies: WebSocket, RTCPeerConnection, UserMedia Average javascript knowledge. You should be familiar with browser's...

Requirements1 - AWS Clustering( web application, janus wss, chatting wss) ...calls. Load Balancing & failover with Janus WebRTC server & APIs (Session wise) between room to room. Requirements2 - Janus Recording (Multi User rec, presentation + User, Shared + User) Requirements3 - Video, Audio Error "stack":"Error: Failed to execute 'createOffer' on 'RTCPeerConnection': Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description streams for m-section with mid='1'.."},"error_description":"(OperationError)Failed to execute 'createOffer' on 'RTCPeerConnection': Session error code: ERROR_CONTENT. Session error description: Failed to set remote...

You have to solve webrtc problem in flutter. E/flutter (11583): [ERROR:flutter/lib/ui/(157)] Unhandled Exception: Unable to RTCPeerConnection::createAnswer: PeerConnection cannot create an answer in a state other than have-remote-offer or have-local-pranswer.

We require an expert on many to many/one to one peer to peer video conferencing implementation of web rtc using RTCPeerConnection implementation in angular/javascript, (and preferably also in android and ios). This should be able to display the video/audio/data with ability to mute/unmute for the organizer of the event. The preferred signalling server is socket io. Suggestions welcome on any other better signalling server also. - video/audio/data should be available to all joined people and all of them should be able to speak - The organizer should be able to mute/unmute the channel - The joiner should be able to swich on/off his own video and also mute/unmute himself This currently an MVP project. The code will be integrated with the actual application later which will be posted...

...on different platform i.e: (from web portal to IOS app, Android app to web portal and so on. WebRTC standard Peer-to-peer, or P2P, describes a connection from a client device to another client device without the use of servers. It's your mobile phone directly connecting to your colleague’s laptop at work, or to your friend’s tablet at home. With the new open web-standard, WebRTC and its RTCPeerConnection API, your web browser has learned how to do just that, so that data transport from one web browser to another web browser is now possible. We will use this standard to enable video call between the client A and client B. It can also be used to share files or enable audio only communication on any device with android, IOS or Web browser. The User module: This ...

I have a working Codeigniter application and I want to add live Video chat over WebRTC. The video chat will only be accessible to authenticated users (this part will not be covered in this project). I need code for the signaling (managing) process involved when working with RTCPeerConnection. The application runs on a MySQL database. I don't need the code to be written for Codeigniter (I can do this myself), just PHP. The code needs to handle just 2 peers at a time. The winner will only be paid after I test the code and it's fully working.

We have a working project using text chat and video chat. We can also include audio. But we cannot dynamically switch an existing RTCPeerConnection from Video to Video+Audio. If you can help us doing the renegotiation like explained in the link below, then you can apply to this role.

I started a module using WebRTC. The clients can chat together using webcam and voice. The owner of the channel share his webcam to every clients. By default, the voice stream is not added. THIS IS ALREADY WORKING. I need to be able to enable the voice with RTCPeerConnection renegotiation such as explained here : But with the current code. I don't have much time, this is why I'm looking for external help.

...24 hours We need to build a CTI Adapter using that connects all the calling functionality data and video with Asterisk and Salesforce You must be having good hands on knowledge of implementing webrtc and 3 main task of acquiring audio and video, comunicating audio and video , communicating arbitary data You must have a good understanding of concepts such as RTCPeerConnection : Audio and video communication between peers RTCDataChannel , RTCSession Description , Stun and Run Very important that you understand the MCU architecture ( Multi Point Control Unit large N-way Call )and the beniits of it over Turn and Stun Architecture Web RTC is ideal to be hosted on Google AppEngine or Amazon Webservices architecting a scalable solution as the number of users increase

...that uses webrtc to run webinars. Module and test site can be found here... The above test site is not my site but rather the module i run on my site. It currently uses... I want it upgraded to this script.. ALL YOU NEED TO DO IS TAKE THE ALREADY DONE SCRIPT FROM RTCMultiConnection AND REPLACE THE RTCPeerConnection CODE. What i am looking to gain from this is. 1. Ability to not only broadcast one-to-many as it works now. But to also run meetings . So I want it to have multiple broadcasters. So a room with up to like 5 on cam and mic and the rest in the already programmed chat room. A.

...that uses webrtc to run webinars. Module and test site can be found here... The above test site is not my site but rather the module i run on my site. It currently uses... I want it upgraded to this script.. ALL YOU NEED TO DO IS TAKE THE ALREADY DONE SCRIPT FROM RTCMultiConnection AND REPLACE THE RTCPeerConnection CODE. What i am looking to gain from this is. 1. Ability to not only broadcast one-to-many as it works now. But to also run meetings . So I want it to have multiple broadcasters. So a room with up to like 5 on cam and mic and the rest in the already programmed chat room. A.

...24 hours We need to build a CTI Adapter using that connects all the calling functionality data and video with Asterisk and Salesforce You must be having good hands on knowledge of implementing webrtc and 3 main task of acquiring audio and video, comunicating audio and video , communicating arbitary data You must have a good understanding of concepts such as RTCPeerConnection : Audio and video communication between peers RTCDataChannel , RTCSession Description , Stun and Run Very important that you understand the MCU architecture ( Multi Point Control Unit large N-way Call )and the beniits of it over Turn and Stun Architecture Web RTC is ideal to be hosted on Google AppEngine or Amazon Webservices architecting a scalable solution as the number of users increase

I have a working Codeigniter application and I want to add live Video chat over WebRTC. The video chat will only be accessible to authenticated users (this part will not be covered in this project). I need code for the signaling (managing) process involved when working with RTCPeerConnection. The application runs on a MySQL database. I don't need the code to be written for Codeigniter (I can do this myself), just PHP. The code needs to handle just 2 peers at a time. The winner will only be paid after I test the code and it's fully working. I will answer any questions over a PM. Please type : "Sochi2014" in the beginning of your bid or message so I know you're not a bot.

...needed - users can post their name, language and country on the main page(kind of wall, one page only at the moment): this posts can be filter by name,language, country people see other on the wall page, can click on the address and triggers the call caller triggers and the receiver answers the redux store has to handle: 1) state of user logged into the signaling server 2) states like RtcPeerConnection, audio video etc (all states needed to verify that the connection between users is set and working) when the connection opens a clock or timer has to appear for both user counting the time opened, would fine if it's possible to set some message alert at some time for example every 30min of opened connection. a screensharing to the other user is required too. I would n...