Webrtc sip uripekerjaan

...copywriter/social media content creator pentru a se alătura echipei noastre într-o campanie de promovare pe termen scurt, cu posibilitatea de extindere, a unor cărți și instrumente de e-learning destinate tinerilor între 18 și 24 de ani care aspiră să devină polițiști. Responsabilități: - Crearea de text/script pentru 14 postări pe săptămână pentru Facebook și Instagram (inclusiv story și reels) și 6 TikTok-uri, cu focus pe produsele editurii - Conținutul trebuie să fie informativ și atrăgător - Utilizarea creativă a textelor, emoticoanelor și formatarea (bold, italic etc.) Oferta: - Colaborare inițială pe o perioadă scurtă, cu posibilitatea de prelungire până după examenul scris din jumătatea lunii iunie, în funcție de performanță. - Flexibilitate...

...tinerilor între 18 și 24 de ani care aspiră să devină polițiști. Editura noastră oferă resurse educaționale pentru admiterea la școlile de poliție, incluzând un curs gratuit de meditații la legislație, oferit exclusiv membrilor grupului nostru de WhatsApp. Responsabilități: - Crearea de text/script pentru 10 postări pe săptămână pentru Facebook și Instagram (inclusiv story și reels) și 4 TikTok-uri, cu focus pe informarea și atragerea tinerilor la Cursul de legislație; - Conținutul trebuie să fie informativ și atrăgător, cu accent pe utilitatea cursurilor oferite și avantajele participării la acesta. - Utilizarea creativă a textelor, emoticoanelor și formatarea (bold, italic etc.) Oferta: - Colaborare inițială pe o perioadă scurtă, cu posibilitatea de prelungi...

...seasoned developer with experience in WebRTC and Flutter. The perfect candidate will be tasked with creating a versatile video meeting mobile application, supporting both iOS and Android. Key requirements include: - Experience in creating both one-on-one video meetings and group video meeting applications. Your responsibility will be the development of an application that can smoothly handle both. - While it's crucial to have decent real-time video quality, it doesn't have to be top-notch. Nevertheless, the performance should be reasonable, allowing for effective communication. A proven track record in cross-platform mobile application development, particularly using Flutter, is required. Your portfolio should show your previous work with WebRTC or similar ...

I am in need of a highly capable and experienced iOS developer with demonstrated experience with the Linphone app. Key responsibilities: - You will be responsible for implementing major modifications to the existing Linphone app. - The focus will be to enhance communication services, so familiarity with VoIP and SIP protocols is necessary. - As I have a set deadline, timely completion is critical. Ideal candidate skills: - Proficiency in iOS development - In-depth knowledge of the Linphone app - Experience in enhancing communication services on VoIP apps - Demonstrable ability to deliver on tight deadlines Your experience in modifying the Linphone app, enhancing communication services and delivering within tight deadlines will be a great asset to this project.

I've been encountering issues during the deployment process of my app on Cloudflare Pages. Specifically, a build failure error surfaces whenever I try to deploy. Invalid prerender config for /[uri] ⚡️ Invalid prerender config for /_next/data/E2QK4OzDZR-6Ocf_NUpkt/[uri].json ⚡️ ERROR: Failed to produce a Cloudflare Pages build from the project. ⚡️ ⚡️ The following routes were not configured to run with the Edge Runtime: ⚡️ - /[uri] ⚡️ - /_next/data/E2QK4OzDZR-6Ocf_NUpkt/[uri].json ⚡️ ⚡️ Please make sure that all your non-static routes export a config object specifying the edge runtime, like: ⚡️ export const config = { runtime: 'edge' }; ⚡️ I applied this export const config = { runtime: 'edge' }; b...

I've been encountering issues during the deployment process of my app on Cloudflare Pages. Specifically, a build failure error surfaces whenever I try to deploy. Invalid prerender config for /[uri] ⚡️ Invalid prerender config for /_next/data/E2QK4OzDZR-6Ocf_NUpkt/[uri].json ⚡️ ERROR: Failed to produce a Cloudflare Pages build from the project. ⚡️ ⚡️ The following routes were not configured to run with the Edge Runtime: ⚡️ - /[uri] ⚡️ - /_next/data/E2QK4OzDZR-6Ocf_NUpkt/[uri].json ⚡️ ⚡️ Please make sure that all your non-static routes export a config object specifying the edge runtime, like: ⚡️ export const config = { runtime: 'edge' }; ⚡️ I applied this export const config = { runtime: 'edge' }; b...

...to create an extension that adds live broadcasting capabilities to an existing WEBRTC-based video conferencing system from Github. This extension will be specifically designed for my PHP-based social networking site. In addition to these features, the live broadcasting system must function seamlessly with the existing structure of my PHP social networking site: 1. User Authentication for Broadcasting: The broadcasting system should allow user authentication to secure the broadcaster's channel. 2. Push Notifications for Live Streams: When a host initiates a live broadcast, a push notification should be delivered to followers or subscribers. Ideal freelancers for this project have a solid grasp on PHP, WEBRTC broadcasting systems, and social networking platform int...

Descrierea proiectului: Site (brand personal) pentru o profesoara de germana. Obiectivele proiectului: • Scop: pentru a vind...1. Romanii din DE care vor sa ia examenul de: a. Atestat nivel B1 limba germana b. ORIENTIRUNGKURS – cunoasterea statului, a societatii si a istoriei Germaniei 2. Romanii din DE sau RO care isi doresc sa invete limba germana la nivel B1 Cerintele proiectului: • Design-ul site-ului sa fie doar alb, negru, auriu • Professional • Minimalist Alte mentiuni: • Alte informatii, documente, video-uri si link-uri de Facebook au fost primite pe email. • Servicile oferite de Freelancer (deliverables): • Webdesign (pg 1-13) - 150 EUR (factura curenta) • Webdesign (pg 13-26) - 150 EUR (factura urmatoare)

I need the most simple Softphone/SIP app with my logo to use or another reseller voip/phone number provider so we can make calls in/out. I need this for Android, IOS, and PC/Mac. Simple app to call in/out. I need a custom softphone app (cross-platform) developed urgently. The app should be a simple dialer for both incoming and outgoing calls, and should be connected to servers. It's important that the app displays our company logo prominently. Key Requirements: - Development of a cross-platform softphone app - Integration with servers - Basic dialer functionality for both incoming and outgoing calls - Display of our company logo Ideal Skills: - Experience in softphone app development - Proficiency in cross-platform development - Ability to integrate with

Cautam programator cu experienta in crearea de site-uri custom si wordpress. Dorim sa colaboram doar cu o persoana serioasa si cu experienta in programare pentru angajare sa colaborare pe proiect. Oferim si solicitam seriozitate si rapiditate in dezvoltarea proiectelor. (Romanian programmers Only)

I'm in need of a SIP and Asterisk expert to help troubleshoot issues with a SIP interconnect between two companies, focusing purely on voice communication. Key responsibilities include: - Identifying issues within the existing setup - Repairing and configuring the SIP and Asterisk system Ideal skills and experience include: - Proven experience with SIP and Asterisk systems - Strong troubleshooting abilities - Understanding of VoIP technologies - Excellent communication skills

...communication using golang WebRTC Pion. This application is targeted for Desktop and IoT devices. The software should be designed to operate on low-resource ARM-based hardware, such as the Raspberry Pi, with the aim of capturing RTSP streams from an IP camera and relaying them to clients via WebRTC. It will implement a socket-based protocol to communicate with an existing Node.js middleware, which will interface with the software developed in Golang. We currently read RTSP stream and manage WebRTC gathering. But there are some connection issues and improvement to do. Key Features: - Support for Selective Forwarding Unit (SFU) to handle communication Ideal candidates should have: - Proficiency in Golang programming language - Extensive experience and understandin...

...will give you: 1. Virtual Numbers 2. SIP Trunk server I am in need of an expert who can develop a SIP Trunk server with the capability of enabling live voice stream access. Key Requirements: - SIP Trunk Server: The primary purpose of this project is the development of a SIP Trunk server. It is essential that the chosen freelancer has experience in developing such systems. - Live Voice Stream Access: The server must offer functionality to enable live voice stream access. This feature must be robust, reliable, and seamlessly integrated into the server. - Access Control: I need the live voice streams to be secured with user-based permissions. This ensures that only authorized users can access the streams. Integration: - The primary platform that the SIP...

I need a poster for my event. It's like a "Balkan party". Pictures to be used are in the attachment. NO OTHER PICTURES CAN BE USED DUE TO COPYRIGHT!! Look on theese profiles for examples on how those kind of posters are made: So it has to be ve...420x280 px 5. 1920x1080 px 6. 1080x1080 px 7. 1080x1920 px Text to be written: ROBERT BIJELIĆ & DINNA (theese are main characters) DJ DEYVID (this has to be smaller since this is only to support main characters) PETEK (this is when event starts) 17.5. Klub ODEON (this is location) Nova Gorica 16+ (this has to be somewhere indicating that only 16 y.o. and older can enter) Vrata se odprejo ob 21. uri. (this indicated when doors will open - can be smaller somewhere)

I am looking for an Asterisk/SIP expert to help me configure a SIP trunk. Specific tasks include: - Configuring the SIP trunk on my Asterisk server - Assisting with troubleshooting any issues that arise during the setup process Requirements: - Expertise in Asterisk and SIP configuration - Experience with setting up SIP trunks - Familiarity with troubleshooting Asterisk setups Additional information: - I already have a SIP provider - Ongoing maintenance assistance will be required after the SIP trunk has been configured.

We have a voip app, voice calling with sip and fcm setup for the push notifications, we like to have a call alert like whatsapp calling alert that can handle both foreground and background states in Android and iOS. Also mention if somebody already worked on this type of feature, bid and connect to show a demo. Feature Required: Call Notification Incoming Screen Alert on App background and foreground State both killed or behind other apps. When you accept the call redirect to the ongoing call screen and call get connected with sip.

...that handles bulk minting in a single transaction. Now, I would need your help in doing the following tasks A- Uploading to IPFS: We have around 6k images. Out of which we would like to deploy 1K now and upload the rest of images for the collection later during our next sale phase. For IPFS pinning services, we would like to go with the Third Web - Here, I know that we can update uri for uploading the remaining images later for the remaining Please let me know if it would affect anyhow to the buyers who purchased NFTs before uploading these images. B - Regarding deployment on Testnet - Yes, before deploying the contract on Mainnet, we would like to check things on Testnet. Now, please let me know if there is any safe method of deploying on testnet with real json and real ...

...from ongoing voice communication. Upon recognition of specified words, the code will need to send DTMF commands via a SIM7600 or SIP/VOIP system. In order to accomplish this project's requirements: - Proficiency in Python 3.5 is a must, as the script will need to be written fully in this version of Python. - Familiarity with serial communication is key, as this will be the method through which the script interfaces with the SIM7600 or SIP/VOIP system. - A thorough understanding of Google Cloud Speech-to-Text API is essential given its integral role in detecting specified words from speech in real-time. - Experience with telecommunication platforms, particularly SIP/VOIP systems and SIM7600 modules, will be highly beneficial for successful completion of this ...

...the NGINX level and stay using my website using HTTPS relying on GCP-CertificateManager at LB level. How my configuration should look like to accomplish this? This the configuration I trying to use with my Django app. ########### server { server_name _; listen 18000; access_log /var/log/nginx/; error_log /var/log/nginx/; location / { try_files $uri u/proxy_to_app; } location u/proxy_to_app { #proxy_set_header X-Forwarded-Port $http_x_forwarded_port; proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for; proxy_set_header Host $host; proxy_set_header X-Real-Ip $remote_addr; proxy_redirect off; proxy_pass ; } } ########### There is a service I have that uses the same

I'm currently in need of a tech-savvy individual to set up the VLAN on my managed TP-Link switch for configuring the SIP line from Tata. Key responsibilities include: - Setting up the VLAN on the switch . - The task will be performed virtually; access will be given using Anydesk. Ideal freelancer should: - Have extensive knowledge of managing TP-Link switch for VLAN configuration - Be proficient in understanding networking concepts related to VLANs

...user looking for an expert in Odoo customization and webRTC integration to help launch my project. We are a group of developers, we need you to help and guide and train us about Odoo webRTC so we can integrate our api. We need a webRTC expert and Odoo expert to give us guide and assistance of how to integrate our AI model into the Odoo 16 webRTC. We have already build the AI model, the webservice and endpoint to commnicate with it and everything related to the AI. So far we have a module that connects with our AI server, everytime an odoo channel is created it reviews the messages, sends it to the AI and creates a new message in the channel that acts as a reply Tasks include: - Customizing Odoo to fit our specific needs - Integrating webRTC within...

We are a group of developers. We need a webRTC expert to give us guide and assistance of how to integrate our AI model into the Odoo 16 webRTC. We have already build the AI model, the webservice and endpoint to commnicate with it and everything related to the AI. So far we have a module that connects with our AI server, everytime an odoo channel is created it reviews the messages, sends it to the AI and creates a new message in the channel that acts as a reply To achieve this we follow the next structure: 1) redefine create on 2) send message body to IA 3) get message and post a message on the channel So far this works on internal chats, but we want this to work on live chat sessions, the idea behind this is that the clients communicate with the IA With the current cod...

Urgent Loocking one program for change voice in live call / Need this program working on windows for softphone (zoiper ,xlite , eyes beam ) Its will be for call center ! Sip Agent need to change his her voice immediatly and voice must be natural . Girl to man /man to girl Im pay direct 30 usd who find me this program . And propose direct on test .

...efficiently. - The endpoint devices that will connect to the Asterisk PBX are Mobile Operator issued SIP trunks. Ideal Skills and Experience: - Prior experience in setting up and configuring Asterisk PBX systems is essential. - Proficiency in handling and routing calls effectively. - Knowledge of SIP trunks and mobile operators' systems would be a plus. Specific Requirements Requirement: A simple Asterisk PBX installed on our server. It is a voice transit facility with incoming and outgoing SIP trunks. All incoming caller ID’s will be replaced by the DIDs provided for each MNO trunk. Connections 1) Multiple incoming customer Accounts from Soft switches like VOS3000 2) Multiple Outgoing SIP trunks provided by local MNO’s DID assignment/Man...

Hi Mohammed S., We have been in touch regarding PBX some time ago. I need a simple PBX installed that can recieve calls from VOS3000 and route them to SIP trunk provided by operator. The incoming caller I D to be modified to match DIDs provided by the SIP trunk. Also ability to defibe

We have a VoIP app in flutter, voice calling with sip and FCM setup for the push notifications, we like to have a call alert like WhatsApp calling alert that can handle both foreground and background states in Android and iOS. If somebody already worked on this type of feature, bid and connect to show a demo.

I'm looking for someone experienced with Asterisk to help me set up a SIP server for educational and testing purposes. The SIP server will be used with a Jio SIP trunk. Key requirements: - Configure Asterisk as a SIP server on the operating system of your choice - Set up a Jio SIP trunk - Create a demonstration of simple dialing using an open source SIP client Ideal skills and experience for this project: - Proficient in Asterisk server configuration - Experience with setting up SIP trunks - Strong knowledge of open source SIP clients - Good communication skills to help guide me through the setup and demonstration process. My budget is not very high ..

I have Patton Smartnode which gets SIP line in (works) and needs to output to its 4 BRI ports to connect to classic PBX (Avaya IP Office 500). SIP line works BRI Configuration to connect Patton and PBX not, this is where i need help for.

We have a VoIP app in flutter, voice calling with sip and fcm setup for the push notifications, we like to have a call alert like whatsapp calling alert that can handle both foreground and background states in Android and iOS. If somebody already worked on this type of feature, bid and connect to show a demo.

I require a competent WebRTC developer to build a screen sharing solution for me. This system should be designed for desktop use only, focusing primarily on the ability for one user to share their screen with a single other user. Requirements: • No audio is required • No chat session is required Ideal Freelancer Skills and Experience: • WebRTC expertise: The successful candidate needs to possess WebRTC proficiency, having previous experience in building screen sharing functionality. • Cross-browser compatibility: Must have knowledge about cross-browser compatibility issues and ways to work around them. If you meet the skills and experiences stated above, please feel free to tender your bid.

...with the WebRTC requirement of H.264 support. Functionality The functionality demonstrated in the above diagram is broken out to: Ingest Pipeline Responsible for connecting to the IP camera over the RTSP protocol using the rtspsrc element. Once connected the pipeline needs to prepare the stream for consumption by lower pipelines as below. Segment Pipeline Take the input stream and output HLS segment files which will be persisted to disk over NFS. It is important to note that we need a hook so that when the files are written we can send a notification message over the message broker announcing the segment file size, name, etc. Transcode Pipeline This child pipeline, when activated, is responsible for transcoding the input H.265 stream to H.264 to prepare it to be consumed over...

I'm looking for someone to help me set up a SIP trunk on my FreePBX phone system. Key Requirements: - Configuration of SIP trunking: I need you to assist in the setup of a SIP trunk on my FreePBX system. Your expertise in this area is crucial. - Connecting to a specific SIP trunk provider: I have a specific SIP trunk provider in mind, and I need you to ensure that the FreePBX system is correctly connected to this provider. Ideal Skills and Experience: - Proven experience with FreePBX phone systems and SIP trunking: I'm looking for someone who has successfully configured SIP trunking on FreePBX systems before. Please provide examples of similar work. - Familiarity with various SIP trunking providers: Your knowledge of di...

I am in urgent need of an Asterisk developer who can work efficiently on debugging and troubleshooting. Key Tasks: - Fix call dropouts - Resolve audio quality issues - Correct failed call routing The ideal candidate for this project should have: - Strong knowledge of SIP protocols - Experience with Asterisk dial plans - Familiarity with debugging tools and logs Only developers well-versed in these areas need to apply. Achieving solid results in a timely manner is of the essence.

The objective of this project involves the implementation and modification of an open SIP protocol to enable data synchronization between mobile devices and computers. The successful freelancer should possess: - A comprehensive understanding and experience with SIP protocols - Ability to implement and customize existing protocols for specific needs - Proficient in facilitating cross-device connections, specifically between mobile devices and computers - Familiarity with data synchronization methods and applications By the end of this project, we aim to have seamless data synchronization between the devices in question. As such, creative problem-solving skills and a keen attention to detail will be invaluable.

I am setting up a call center and need assistance activating SIP via VPN on Mikrotik OS. The SIP server for the call center will be accessed through a site-to-site VPN using IPsec. Key requirements include: - Configuring Mikrotik OS for SIP calls - Establishing secure site-to-site VPN using IPsec - Ensuring the SIP server is accessible through the VPN - Preparing the system to handle more than 50 concurrent calls at peak times Experience and Skills: - Expertise in Mikrotik OS - Proficiency in SIP server configuration - Strong understanding of VPNs and IPsec - Prior experience with call center setups would be advantageous - Ability to design systems that can handle high call volumes is essential Please include any relevant experience or examples of sim...

I am looking for an experienced professional to install a SIP trunk server on a Linux system. As part of the project, you should also set up access to the live voice stream with the primary purpose of real-time transcription. Key Requirements: - Install and configure a SIP trunk server on a Linux system. - Enable access to a live voice stream for real-time transcription. Ideal Candidate: - Proficient in Linux server administration. - Experienced in SIP trunk server installation. - Familiar with real-time transcription processes. - Skilled in setting up live voice stream access. - Strong problem-solving and troubleshooting abilities.

braodcasting jitsi webrtc Bunlara hakim biri lazım bana

I'm looking for a skilled PHP developer to create a script for my website...script is to enable real-time screen sharing from the website to a TV. Key Requirements: - The PHP script must be able to support screen sharing in real-time. - The script should be able to facilitate sharing between the website and a TV seamlessly. Ideal Skills: - Proficient in PHP, with a solid understanding of web applications and databases. - Experience with screen sharing technologies such as WebRTC, Miracast, or Chromecast would be beneficial. - Strong communication and problem-solving skills to ensure a smooth implementation of this feature. I'm open to suggestions on the specific technology to be used for screen sharing, as long as it meets the requirements and can be integrated effectiv...

I'm looking for a proficient web developer to create a cutting-edge web app tailored for educational institutions. The main features of the app should revolve around seamless video communication between users. Key Features: - One-to-One Video Calling: Implementing a robust video calling functionality that can support both one-to-one sessions. - Live Chat: A real-time chat room integrated within the app for additional interaction. - Screen Sharing: Facilitating screen sharing among the participants. Customization Level: - The primary focus of this project is on basic video call functionality. No need for advanced features such as filters, effects, or custom backgrounds. The emphasis is on providing a stable, user-friendly video calling experience. Ideal Skills: - Proficiency in M...

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

General Event Assistants wanted for a fun, science-themed event. The Lit Lab (@litlablondon) is the UK’s first and only ‘Science and Sip’ experience. In this exclusive, immersive pop-up event, guests will be transformed into scientists to create exciting drinks, conduct science experiments, and compete in challenges against other scientists. Responsibilities: - Act as general event staff for monthly events and some ad-hoc events in London. - Event dates and times TBC. These will be shared at least 1 month in advance where possible, at which time you can accept or decline within 1 week. - Temporary / day work: 4-5 hours work. This is a pop-up event that changes locations. - Setup and clean-up: Moving and arranging chairs, tables, equipment and storage boxes e...

...branding guidelines. Test modifications thoroughly to ensure stability and functionality. Collaborate with our internal team to gather requirements and feedback. Provide documentation and support for the customized software. Requirements: Strong proficiency in Visual C++. Experience with software customization, particularly with open-source projects. Familiarity with VoIP technologies and SIP protocols is a plus. Excellent problem-solving skills and attention to detail. Ability to work independently and manage time effectively. Project Details: This is a freelance, remote position. Duration: Approximately 1-3 months, depending on the project scope and requirements. Payment will be negotiated based on experience and the specifics of the agreed-upon deliverables. ...

I need an experienced professional to help fix an urgent problem in my Asterisk system that utilizes both Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP). My issue revolves around tenant to tenant recording or IVR. Specifically, all audio plays for only 2 seconds before abruptly ending calls. - Skills and Experience You should have extensive experience with Asterisk, SIP, and RTP. A deep understanding of their inner workings and potential pitfalls is necessary to effectively troubleshoot and resolve the current issue. A track record for quick and efficient problem solving is essential. Deliverables: - Diagnose the issue causing the audio and calls to end after 2 seconds - Implement a reliable solution to fix the issue Note that this project deman...

I am looking for an expert who can help me set up a SIP trunk server using Twilio as the SIP provider. This server will be used for both voice calling and SMS messaging. Key Responsibilities: - Establishing a SIP trunk server with configurations optimized for Twilio - Implementing call management features like Call Recording, Call Forwarding, and Interactive Voice Response (IVR) - Ensuring seamless integration of both voice and SMS capabilities - Providing recommendations for server security and performance optimization Ideal Candidate: - Extensive experience with SIP trunking, particularly with Twilio - Proficiency in configuring and customizing SIP trunk servers - Proven track record in implementing call management features such as Call Recording, Ca...

...ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP t...

...ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP t...

I need a MEAN stack developer who has very good knowledge in node.js and angular . My project is at ending stage . I need developer to fix issues and help in process to make application live . It will be a long term project if developer proves to be worthy . My application has following webstack - Node.js -Express -MongoDB -Angular It would be nice , Person has a good working experince in WEBRTC , chrome extension , powershell The project is time-sensitive and will require a dedicated developer to assist in expediting the process of completion. Ideal Skills: - Strong experience in full-stack development with the MEAN stack - Ability to prioritize tasks and deliver results under tight deadlines - Excellent communication skills for collaboration and updates on project pro...

Hi, I need an expert Python developer who has experience with WebRTC. I want to forward incoming/outgoing calls from HTTP to WebRTC and WebRTC to HTTP. Only bid if you are confident and have relevant experience. Thanks

...calendar interface, setting specific times for availability or choosing to accept fan requests on-the-fly. 3. Bidding Mechanism: Fans can bid on available time slots if the influencer enables a bidding option. Alternatively, they can book directly at a set fee if the influencer has predefined their availability and fee. 4. Real-Time Communication: Integration of video communication (preferably using WebRTC technology for real-time, high-quality video chats). 5. Payment Integration: Secure processing of payments with options for fans to pay for sessions and influencers to receive earnings, minus a platform commission. 6. Responsive Design: The app should be fully responsive and functional on both desktop and mobile devices. 7. Security Features: Ensuring data security and privacy ...

I have developed a VR(Meta Quest, using Unity ) application and need to interact with it via a Smartphone App. I have developed a basic app containing WebSocket server to manage communication with Flutter but the following functionalities need to be implemented: - Casting using WebRTC: The VR side functionality has been implemented. The App side functionality needs to be created. - Autodiscovery and auto-connect via local network: The App and VR have to be able to connect to each other automatically without the user having to set manually the IP address. - Security: Implement security protocols for the VR- Smartphone app interaction. - UI and UX improvement: Basic UI is implemented but needs improvement There will be a follow-up opportunity to improve the application upon suc...