Webrtcpekerjaan

need to develop a Webrtc app for Android using Xamarin Forms

need to code a webrtc for Android in Xamarin forms. msg to discuss in detail

I would like a multi-class WebRTC tutorial, at first with the most basic concepts (from zero) and increasing the level with each class, starting to beginners and finishing with a professional mastering of the subject, to learn and understand completely how working RTC with APIs. The minim total duration of the videocourse with all of classes is 2 hours.

We would like to use your experienced help for our website by using nodejs and vue/nuxt. You must be passionate about good quality and willing to show your own real work. Making UI with Webrtc and VOIP may be a good skill.

...of PHP, we had to point my payment processor "Stripe" to another server with the latest version of PHP. I need my Stripe account configured to work on this new server I will be using. You will need to look at the Stripe (payment processor) code to see what I am referring to. Server has a Plesk Control panel and is a dedicated server. Task #3 - I want to upgrade my current video chat feature to WebRTC. It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins. Right now, I'm having to use a third party for my video chat. Whoever I hire I would like for them to do more work and customizations on the site eventually. Let's see how this job goes first. I can pay you di...

Proyecto WebRtc universidad . App ionic

Video Conference Web Solution We want to centralize several reception of a certain company in a central post. The solution is almost ready, we just have instability in the connection! The idea was built on the WebRTC API for connecting peers. The system is having trouble establishing the connection between the points. The problem is in the query to the stun server, which in many cases takes a long time to respond with the connection parameters. At first the idea is to create a stun server with the CORTUN lib. CORTUN is an open source server.

Busco experto para integración de app en Janus y soporte técnico.

Im looking for experienced devs/electronics to build very low latency video streaming system for a drone. Stream P2P over Internet. Latency glass-to-glass should be less than 100ms + network. If you got REAL experience with webRTC ping me for more info. It is a proof of concept only project.

Git repository: We have a mini project in laravel 5.6 and react . We would like to upgrade the project to Laravel 8 and migrate React code to Vue.js. Github link: Steps to complete the project: - create a Github public repository and move the code there; - create a live dev environment link to track the progress - if necessary we will provide a VPS machine ( you have to install Apache, PHP, Mysql, Laravel by yourself); - Upgrade Laravel 5.6 to Laravel 8+ (PHP 7.4 or 8) - Migrate react to latest vue.js The functionality must remain the same. We will pay only for the working version. Please follow the tutorial on youtube: Make

Required experience: - Highly experienced with GStreamer (C++) - Mandatory - WebRTC experience (major advantage) - FFMpeg or other solutions that may provide what we need (advantage) 2. Video receiving application that implements following GStreamer pipeline: a. Receives data from step 1.f above b. Sends data into appsink for custom processing: i. A simple placeholder routine that decodes video frame - pull sample, get buffer, decode content to new buffer and push buffer. c. Application sends RAW data from 2.b to appsrc d. Appsrc sends data to (configurable): i. v4l2sink ii. x264enc that: 1. Encodes RAW data 2. Sends it to webrtcbin e. Full pipeline is: udpsrc -> appsink -> custom processing -> appsrc -> v4l2sink or x264enc -> webrtcbin 3. Implementation details: a. ...

У меня на хостинге стоит скрипт видеочат основан на flash, технология устарела, перевести или сделать с таким же интерфейсом на webrtc! Flash сейчас не поддерживается, то необходимо использовать современные технологии. Ранее у меня был Flash который стримил через rtmp. Сейчас для связи p2p люди используют webrtc - это более простое решение(с точки зрения разработки) и оно идеально, мне кажется, подойдет . Мне нужно ) создать видео чат на webrtc и поместить это все в движок, вместо схемы с Flash и RTMP. Можете помочь *? Может быть есть какое -то еще решение данного проекта !

У меня на хостинге стоит скрипт видеочат основан на flash, технология устарела, перевести или сделать с таким же интерфейсом на webrtc! Flash сейчас не поддерживается, то необходимо использовать современные технологии. Ранее у меня был Flash который стримил через rtmp. Сейчас для связи p2p люди используют webrtc - это более простое решение(с точки зрения разработки) и оно идеально, мне кажется, подойдет . Мне нужно ) создать видео чат на webrtc и поместить это все в движок, вместо схемы с Flash и RTMP. Можете помочь *? Может быть есть какое -то еще решение данного проекта !

Data parsing communication between two WAN site without port forwarding. Web real time communication video conferencing also required

I do have coturn server on ubuntu 20.4 installed by a developer to serve Bigbluebutton server which is installed on a dedicated server, and also Junus webrtc server which is installed on the same server that coturn server installed on. The person who have installed said that he has configured the stun server but he couldn't get the turn server to relay the traffic. I need to configure the coturn server to serve these servers. I do have only 1 IP address but each of these servers has its domain name and I am using ports 80 and 443 for SSL for the Bigbluebutton server and Let the Nginx on that server be a proxy for the rest of the servers MAX bid $50

...selenium or a browser that you can most suitable bidders will be evaluated. Features we want to change in the browser: Canvas data, fingerprint Webgl data fingerprint. Video card properties Audio settings font list. plugin list. Platform id Touch support Battery capacity Gamepad USB devices list SVG reading Headers order Connection, should get proxy ip value when proxy is used. Webrtc ip should get proxy ip value when proxy is used. Display specs – resolution, color depth, etc. All explorer properties should be changed by default: userAgent, oscpu, vendor, vendorSub, productSub, buildID, HardwareConcurrency, deviceMemory and so on: getGamepads and getVRDisplays. Smart change of browser language - language format should be changed according to proxy when using proxy Geolo...

Needs to integrate Audio/Video/Chat System in Consultation Mobile Kotlin App via WebRTC - VoIP Complete functionalities need to be implemented along with Server setup Needs to be implemented in all required new screens in mobile. i.e. Audio/Video/chat , Audio Video switching, Call Waiting, Disconnection, Data Saver, encryption/Decryption...all basic functionalities for Mobile audio/video one line consultation. Support for all your video use cases: 1:1 consultations, group video chat, screen sharing, and large scale broadcasts to thousands Recording of any session, plus vital data, including duration and quality of every session and stream Advanced security, firewall-control, regional isolation, and compliance certification options Technology - Android Kotlin

Would need someone to fully install Android Studio on a virtual machine (preferably on Windows) with WEBRTC, Java, etc... to be able to correctly edit and build "" and "" apps.

I am a professional unreal engine developer having experience in different aspects of unreal engine software development. I develop games, simulation systems, aws hosted dedicated multiplayer servers and much more. ONLINE Multiplayer CONCERT SYSTEM (AWS) with Pixel Streaming WebRTC people were able to join together and enjoy live concert in a Open World map, with high quality 3 stages where different artists appear at their timings. People must be able to buy passes for extra features, joy Rides and much more. Must be setup with desktop software must be on Cloud and pixel streamed so that any user with a browser can play and enjoy it.

Requirements: 1. Raspberry pi would be connected to a keypad, mic, speaker and camera. It would have a build in MySQL / MariaDB database. When a user presses a number, the application should log in mysqldb, collect the corresponding mobile numbers stored against that number and dial out / make a video call to those mobile numbers simultaneously / group call. The application should further record the call as a video file in raspberry pi 2. The receivers shoudl have a mobile app (Android / iOS). The receiver should have the provision to accept or reject the call. Once the call is accepted, then the receiver should receive audio and video from the raspberry pi, but should send out only audio. After the end of the call, there should be two button provided in the app, which should menti...

Live video streaming and broadcast over a custom application using Azure Media Services.

Hello I want to to create WebRTC widget that can run on Mac. Widget should able to drag. It's simple app, I will give you detail.

Looking for a developer for a video call/chat app. With skills in XMPP, WebRTC, SIP

Google / Alexa Smart home skill webRTC Camera setup.. need to build out a custom skill to enable webRTC over the web browser. Need someone that understands the webRTC setup offer/answer Both dont support trickle ICE though google states its half duplex audio where Alexa is full duplex

Required experience: - Highly experienced with GStreamer (C++) - Mandatory - WebRTC experience (major advantage) - FFMpeg or other solutions that may provide what we need (advantage) 2. Video receiving application that implements following GStreamer pipeline: a. Receives data from step 1.f above b. Sends data into appsink for custom processing: i. A simple placeholder routine that decodes video frame - pull sample, get buffer, decode content to new buffer and push buffer. c. Application sends RAW data from 2.b to appsrc d. Appsrc sends data to (configurable): i. v4l2sink ii. x264enc that: 1. Encodes RAW data 2. Sends it to webrtcbin e. Full pipeline is: udpsrc -> appsink -> custom processing -> appsrc -> v4l2sink or x264enc -> webrtcbin 3. Implementation details: a. ...

making video conferencing call between max 6 devices using webrtc, python, raspberry pi at one end and mobile at the other ends.

We are looking for a webrtc developer who has great experience in creating Audio / Video streaming app web base. Must have extensive experience in this field

We are looking for Full-Stack Developer for our project. Matters of the site and project will be discussed with the final candidate as it is a confidential project. Full stack/ Back end technologies: - PHP 7.x/8 - MySQL Knowledge: - Knowledge of Digital Ocean, firebase, aws - Payment system API integration - Chatbots - WebRTC experiance

We need a WebRTC client SDK that can be implemented in 3rd party projects. Needed functionalities: WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/disable video during a session (during an ongoing call session re-INVITE and disable or enable video) WebRTC signaling plane: - SIP over WebSecureSocket (will connect to a sip server as Kamailio/Opensips/FreeSWITCH) WebRTC media plane - codecs: 711, opus, VP8, VP9, H.264 - DTLS/ICE/SRTP API facing side: - provide an easy and comprehensive API for quick integration into 3rd party projects

hi we have a web conference project built with nodejs - on backend and javascript on frontend. for the voice, video and screenshare transmission we need to use a media server like kurento or mediasoup. I should say that our preference is using mediasoup.

We need a developer who can create a WebRTC server using Janus, JSSIP, WebRTC, mediasoup, etc. (whatever solution is best). The system works in two ways: 1 (attached graphic). Calls will come in through our DID. Each call is authenticated through our API (handled by a separate developer) and then each user is placed into separate channels after API authentication. The audio stream from the caller will be listened to and processed by our audio tools. The calling system is setup through Twilio SIP trunking. 2. When each user is in a webrtc channel, allow for the ability to have one or two users join via webrtc clients such as our mobile app, browser, etc. (another developer will handle client development). We are NOT building a conferencing app, but we need the ...

Im looking for experienced devs/electronics to build very low latency video streaming system for a drone. Stream P2P over Internet. Latency glass-to-glass should be less than 100ms + network. If you got experience with webRTC ping me for more info.

Looking for an experienced webRTC developer who can integrate "Agora video solution" to our site

We are looking for an experience company or a team capable of development and future support of a custom Kazoo PBX. Experience Requirement: -Must have built something similar and demo the solution with Kazoo -Previous development with Kazoo, Freeswitch, Kamailio -Examples of previous development of teleph...understanding of Database (SQL) with High Availablity (HA) & data replication/sharing, DRDB IF YOU HAVE NOT DESIGNED AND DEVELOPED A SIMILAR SOLUTION, do not apply. we require an experienced team capable of full delivery on time and within budget. The initial scope of the project will include creating the following options -Visually designed IVR option tree -Call Queue system -SMS Implementation with WebRTC -Building Mobile Apps (iOS and Android) -Building a desktop app (...

We are looking for a WebRtc developer who has great experience in creating Audio and Video streaming services. Must have extensive experience in this field .

We need developers on skill webrtc voip mediasoup sip (Budget: $250 - $750 USD, Jobs: Java, SIP, VoIP, WebRTC)

The current application has video conferencing capability using the Kurento media server, and the goal of this minor enhancement is to add screen sharing using Kurento's client-side library. It should be straightforward for someone with experience. We can provide support on the signaling side if needed.

i need a developer to setup a simple turn and stun server procedure for my tsp stream or any other suggestions are also welcome. Where i require to bypass the firewall and send the video stream to remote server for further processing.

We need developers on skill webrtc voip mediasoup sip ejabberd

Both user on the same intranet show be able to make call locally with calling external API You are to use signalr .net core 3.1 Razor view WebRTC

Both user on the same intranet show be able to make call locally with calling external API You are to use signalr .net core 3.1 Razor view WebRTC

Create a application in which we have option for video chatting. Any number of users can join video conference. The application should have option to call multiple user at a time. Group Calling functionality: 1. The application should have the group list (under each group, any no of users will be available) 2. User can select any of the group a...group list (under each group, any no of users will be available) 2. User can select any of the group and could make call 3. Calling screen should be displayed for all the users under the group. 4. When the users attend the call, the users streams should be available for the called user and for all the other attended users (as in Whatsapp) Note: Third party tools should not be used. Try to use webrtc and peerjs packages available for rea...

(IMMEDIATE REQUIREMENT). We need a video calling feature in our app, we choose the AppRTC (webRTC) by google to use as video calling service. We need to open the appRTC URL on in app browser which will take 1 param (Room id), and starts the video calling.

My project working fine in locally but after hosting not working properly so i need to fix that issue.

webRTC janus-gateway SDP noSIP /sip plugin setup.. need to setup a call from a SDP Offer connection.

We have a audio/video call application using flutter and webrtc. We want to add ringing feature to the application. In our app we are using firebase notification for messages and it would be good to use that for ringing feature too.

Google Assistant webRTC developer This will be 2way call SIP / WebRT job

We are an early age-raised startup looking for 4 MERN stack developers who can help us to build a fully functional LMS platform , The devs should have experience in react, tailwind CSS,webrtc . Also, we are looking for one who can automate using the latest technologies. Bid only if you have good experience and share some links to your past projects

we need to install GStreamer on our Amazon Linux machine and make it subscribe to ion-sfu following this example -> you should be highly familar with both GStreamer deployments and Amazon Linux AMI handling in order to subscribe to this working SFU server to capture audio/videos rec...on our Amazon Linux machine and make it subscribe to ion-sfu following this example -> you should be highly familar with both GStreamer deployments and Amazon Linux AMI handling in order to subscribe to this working SFU server to capture audio/videos recived streams. a big bonus is GoLang and WebRTC knowledge here, as ion-sfu is Go WebRTC project.

Looking to Get a project moving with some development updates.