Webrtc sip uri työt

Hi I have a working ipsec tunnel (one host as a client and one host as a server) and I need to make sip calls from asterisk (as a client) to freeswitch (as a server) thru this tunnel. It works outside the tunnel but when I try to use the ipsec tunnel IP I can't hear audio and get call hangup after few seconds. I can provide ssh to both client and server and it's easy to 'switch' between using and not using the tunnel. Please write down the changes that you make. Also please only bid if you have experience with freeswitch. Max 100 euros. Thank you.

I'm seeking a proficient developer to construct a flutter web chat application Build api php back end mysql database and flutter. The app should include three core features: - Text messaging - Group messaging - Media sharing -WebRTC Video call -Back end admin panel Also, users should be capable of creating unique profiles on the platform. Ideal expertise for the project includes strong knowledge of cross-platform development, experience in chat app development, and understanding of user profile creation. Familiarity with designing and implementing media sharing interfaces is a substantial plus.

I'm seeking a proficient developer to construct a cross-platform chat application compatible with iOS, Android, and Web. Build api cross platform php back end apis mysql database and flutter. The app should include three core features: - Text messaging - Group messaging - Media sharing -WebRTC Video call -Back end admin panel Also, users should be capable of creating unique profiles on the platform. Ideal expertise for the project includes strong knowledge of cross-platform development, experience in chat app development, and understanding of user profile creation. Familiarity with designing and implementing media sharing interfaces is a substantial plus.

I'm looking for iOS developer to allow my webview app to have video calls. the video call function is working fine on Chrome. but in the video call app it gives this error "WebRTC" is not available in your browser.

I'm seeking a proficient developer to construct a cross-platform chat application compatible with iOS, Android, and Web. Build api cross platform php back end apis mysql database and flutter. The app should include three core features: - Text messaging - Group messaging - Media sharing -WebRTC Video call -Back end admin panel Also, users should be capable of creating unique profiles on the platform. Ideal expertise for the project includes strong knowledge of cross-platform development, experience in chat app development, and understanding of user profile creation. Familiarity with designing and implementing media sharing interfaces is a substantial plus.

I'm seeking a proficient developer to construct a cross-platform chat application compatible with iOS, Android, and Web. The app should include three core features: - Text messaging - Group messaging - Media sharing -WebRTC Video call Also, users should be capable of creating unique profiles on the platform. Ideal expertise for the project includes strong knowledge of cross-platform development, experience in chat app development, and understanding of user profile creation. Familiarity with designing and implementing media sharing interfaces is a substantial plus.

I'm searching for a proficient SIP/Asterisk engineer who has ample experience in performing the following tasks: - Setting up and configuring an Asterisk server - Ensuring that the server is set up for high security and can support telemarketing or sales calls - Adding a specific function of calling and spoofing numbers, to be used exclusively for telemarketing and sales purposes Ideal skills and experience for the job: - Extensive experience with Asterisk and SIP protocols - Past-experience in setting up IVR systems - Understanding of server security measures - Familiarity with telemarketing systems Please provide a summary of your relevant work experience and your approach to this task in your bid.

I need an experienced Full Stack React Native developer to build an app that is clean and professional in design. The important features I require are: - User authentication: You should be proficient in building secure user authentication systems. - Push notifications: Extensive knowledge in implementing push notifications seamlessly is essential. - WebSocket and WebRTC experience are required. The app should be compatible with both Android and iOS. Familiarity with both platforms is highly appreciated. Proficiency in UI/UX design to reflect a clean, professional aesthetic is a must. Please provide links to apps you've worked on that showcase similar features.

...programming languages. Additionally, you should possess a deep knowledge of social media APIs, including but not limited to Facebook, Twitter, and Instagram, as well as experience in integrating these APIs into applications. Furthermore, a solid understanding of live streaming technologies is crucial for this position. You should be familiar with popular live streaming protocols such as RTMP, HLS, and WebRTC, and have experience in implementing them in previous projects. Your ability to troubleshoot and resolve any issues that may arise during the live streaming process will be essential in ensuring a seamless user experience. Aside from technical skills, we are also seeking someone with strong problem-solving and communication skills. As a developer on this project, you will b...

...proficient full stack developer, experienced in Stripe KYC and APIs to handle payments and construct an engaging admin section for my project. The skills and qualities essential for this job include: - Proficiency in JavaScript, TypeScript, Next JS - Solid experience with Stripe KYC and APIs - Knowledge in planning and creating user-friendly admin sections - Must have experience working with webRTC, websockets, Admin Section Desired Features: - User Management: The ability to control and manage user profiles. - Dashboard Analytics: The ability to track and display crucial data and metrics. - Content Management: A system that efficiently manages and controls website content. Stripe Payment Functionalities: - Handling One-time payments: Implementing a functional and secure sys...

I need a project with essentially two page, in one page i need to record my voice (from my phone mic), in another page (maybe listen page) the customer need to ear the stream of voice recorded in realtime from page one. The customer login is based on pin generated by record session id, also i need to save the data of customer to mysql Db

I require an experienced iOS developer to integrate WKWebViewRTC into iOS webview app. this should enable video call functionality. Key responsibilities: - Install WKWebViewRTC to existing project - Test and ensure smooth operation of video calls here is what need to be done:

...Current audio streaming using: Our app is like a voice club where our user can just create a voice room and a random person can join this voice club. Our app is working well with current audio streaming with Agora. But we don't want to use agora. We want to build our own WebRTC audio streaming in our VPS. So, what I need is I want to replace Agora in my application with a custom audio streaming server. Using Nodejs we will create the audio streaming server using webRTC. And then using express js it will have some route to implement in our android application. Our android studio code is in GitHub: Our Backend Nodejs Code: These are private repos, so I will add you on GitHub

...GPS capabilities: This includes live tracking of users, efficient route planning for navigation, and sending location-based notifications to users. **Skills and Experience Required:** - Proficiency in both Android and iOS app development, with a solid portfolio showcasing similar projects. - Strong experience in implementing real-time communication features, ideally with a background in using WebRTC or any relevant technologies. - Experience in integrating GPS features into mobile apps, including but not limited to live tracking, route planning, and push notifications based on user location. - A good understanding of UI/UX design to ensure the app is user-friendly and intuitive. - Must be knowledgeable in ensuring data security, especially in real-time communication and user loc...

I'm searching for a network security expert to assist in diagnosing a problem with my SIP ALG settings on my Cisco ASA5506 firewall. It's crucial because I am experiencing an issue wherein the phone rings for one second and then cuts off. Your tasks will include: - Examining the SIP ALG status on my firewall. - Troubleshooting any issues found and offering solutions to rectify them. - Implementing the solution with my approval. Relevant Skills and Experience: - Proven experience with Cisco ASA5506 firewalls. - Deep understanding of network security. - Skills in VoIP technologies and SIP ALG. - Familiarity with various VoIP service providers/software is a plus. Please note, we are not currently using Skype for Business or Ring Central, but a different prov...

...Current audio streaming using: Our app is like a voice club where our user can just create a voice room and a random person can join this voice club. Our app is working well with current audio streaming with Agora. But we don't want to use agora. We want to build our own WebRTC audio streaming in our VPS. So, what I need is I want to replace Agora in my application with a custom audio streaming server. Using Nodejs we will create the audio streaming server using webRTC. And then using express js it will have some route to implement in our android application. Our android studio code is in GitHub: Our Backend Nodejs Code: These are private repos, so I will add you on GitHub

I resolved issues with developed pbx based asterisk with dial plan and fax over sip

I will install Freepbx on a local server and in Digital Ocean Cloud. I will configure it for 10 users and 3 trunks, while also ensuring that configurations can be synced between them. Finally, I will secure it against external threats and SIP flood attacks.

I need an imaginative designer to create a logo for my Italian Bistro, SipsAndSpices, where flavor meets elegance. Our bistro offers a curated selection of fine wines, craft cocktails, and globally inspired dishes. With a warm ambiance and attentive service, we invite you to savor the exquisite blend of flavors and sip on carefully chosen libations. I am open to all styles, be it minimalistic, vintage, modern, or a unique blend of these. Color is also your playground; creative use of blue, green, yellow, or any other shades are welcome. Just bear in mind that the logo should reflect an Italian restaurant vibe. Skills and Experience: - Proficiency in graphic design software. - Knowledge of effective color schemes in logo design. - Ability to design within different styles. ...

...{{ssl_certificate}} server_name ; return 301 $request_uri; } server { listen 80; listen [::]:80; listen 443 ssl http2; listen [::]:443 ssl http2; {{ssl_certificate_key}} {{ssl_certificate}} server_name ; {{root}} {{nginx_access_log}} {{nginx_error_log}} if ($scheme != "https") { rewrite ^ https://$host$uri permanent; } location ~ /.well-known { auth_basic off; allow all; } {{settings}} location ~/.git { deny all; } location ~/(wp-admin/|) { proxy_set_header X-Real-IP $remote_addr; proxy_set_header X-Forwarded-For $remote_addr; proxy_set_header X-Forwarded-Host $http_host; proxy_set_header Host $host; proxy_pass ;

I'm seeking an expert Android developer to create a VoIP app with GSM integration. This pro...developer to create a VoIP app with GSM integration. This project includes, but is not limited to: - Developing an app for calling and text messaging through a SIP server, inclusive of delivery reports. - Implementing features such as call recording and instant messaging. - Ensuring compatibility with older Android versions. Since I don't have a preferred SIP server, your input and recommendations would be highly valued. Experience with SIP servers and a strong understanding of both VoIP principles and Android development are necessary for this project. Your role will be pivotal in guiding the selection and integration of the SIP server. Please consider the ...

Salut, Pe lan 1 avem reteaua locala calculatoare si telefoane voip si internet Pe lan 2 avem acces la serviciul digi sip care la care sa conectat dar nu mai vedem telefoanele din retea initial vedeam telefoanele ba chiar am gacut si teste cu un modul GOIP care a functionat bine .

...project. The ideal candidate will have extensive experience with Puppeteer for automating web browsing and data scraping, familiarity with ffmpeg for any potential video editing needs, and solid knowledge of WebRTC for real-time communication and video streaming. The project will focus on two main tasks: 1. **Web Browsing and Data Scraping:** - Automate the process to extract specific information from various websites. - Leverage Puppeteer to navigate, simulate user interaction, and scrape data efficiently. 2. **Real-Time Communication and Video Streaming:** - Implement WebRTC to facilitate online meetings and webinars, ensuring stable and seamless video streaming capabilities. - Possess the technical know-how to resolve common issues related to live video fe...

We are seeking expertise to revamp our current online meeting application powered by WebRTC, aiming for improved scalability and functionality. The application's key components include: A WebRTC server built with Node.js A front-end interface developed in React.js, facilitating the scheduling of 2-party meetings between participants, generating unique meeting IDs, and supporting a browser-based meeting recorder. Notably, our solution is entirely custom-built, devoid of third-party APIs for recording or other functionalities. Objectives for Refactoring/Rearchitecting: Multi-Party Support: Enable the application to support meetings with up to 100 participants, ensuring stability and performance. Independent Audio and Video Recording: Implement separate recording streams...

...capabilities. This project is critical for improving accessibility and ensuring compatibility with current web standards. **Core Requirements:** - Convert Flash chat to a fully functional HTML5 application. - Retain voice and video chat functionalities. - Implement specific design and branding guidelines provided by me. **Ideal Skills and Experience:** - Strong proficiency in HTML5, CSS, JavaScript, and WebRTC (for voice/video chat). - Experience with modern web development frameworks and libraries. - Knowledge in creating responsive and cross-browser compatible web applications. - Experience in working with legacy Flash applications and converting them to HTML5. - Understanding of UI/UX design principles, particularly in the context of chat applications. - Ability to work with...

More details: What type of engineer do you need? Structural engineer Could you please clarify what specific applications of SIP panels are you interested in? Wall system, Roof system, Floor system What are the approximate dimensions of the cottage? Small (less than 500 sq ft) , 12 feet wide by 20 feet long ICF blocks for foundation and walls. Sip panels for floor, Gable ends for roof and roof engineering for attachments See attached file to read details

Buna ziua, Vreau sa dezolt un site / magazin online in platforma opencart . Sa pot implementa feed-ul de la furnizor si sa il si sa pot face export in alte plaforme. Sa aiba aiba link-uri curate, scurte.

...essential configuration changes to my existing Issabel IP PBX setup. The project specifics include: 1. Changing the current SIP provider from Tata to Airtel. 2. Updating IP PBX configuration settings to seamlessly support the new Airtel SIP service. 3. Making necessary modifications to the SIP trunk settings to ensure optimal compatibility with Airtel's service. 4. Configuring additional SIP trunk registrations as required. The ideal candidate should possess proficiency in SIP trunk configurations, with a solid track record of experience in working with both Tata and Airtel SIP services. A clear understanding of Issabel IP PBX setups and the ability to configure SIP settings accordingly are crucial for the success of this project. ...

I'm seeking an experienced C and NodeJS developer to efficiently develop new functionalities and bolster performance on a C-based WebRTC implementation in a remote administration tool. Here are the repositories for this tool: Key areas of concentration will include: - Modify the WebRTC implementation to use a custom screen capture source. Instead of capturing the display buffer the operating system uses, implement a method to capture the screen at a lower level, possibly by directly hooking into the video hardware. This might address the manipulation of the display buffer and the screen capture blocking, as it doesn't rely on the standard screen capture APIs. - Integrate WebRTC's rendering pipeline directly with

More details: References: Which browsers do you need the system to support? Google Chrome, Mozilla Firefox, Safari What information should successful freelancers include in their application? Past work

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...social media platforms to facilitate broader engagement. - **Potential for Future Integration:** While not immediately required, keeping options open for integrating cloud storage services and payment gateways in the future would be a plus. **Ideal Skills and Experience:** - **Node.js Proficiency:** Demonstrated experience in building applications using Node.js is a must. - **WebRTC Knowledge:** Understanding of WebRTC for live streaming functionalities. - **API Integration Expertise:** Skilled in integrating various APIs, with a focus on social media platforms. - **Solid Communication Skills:** Ability to clearly articulate progress and hurdles will be key for smooth project development. This project requires a freelancer who is not only technical but can also foresee...

...Current audio streaming using: Our app is like a voice club where our user can just create a voice room and a random person can join this voice club. Our app is working well with current audio streaming with Agora. But we don't want to use agora. We want to build our own WebRTC audio streaming in our VPS. So, what I need is I want to replace Agora in my application with a custom audio streaming server. Using Nodejs we will create the audio streaming server using webRTC. And then using express js it will have some route to implement in our android application. Our android studio code is in GitHub: Our Backend Nodejs Code: These are private repos, so I will add you on GitHub

I need a comprehensive telephony solution, utilizing both FreePBX and Asterisk servers to streamline incoming, outgoing, and call monitoring processes. Here's a glance at my requirements with the BSNL SIP Direct Truck : 1. FreePBX AND Asterisk Server Setup: I am looking for an experienced tech consultant who can not only advise but also assist in setting up both FreePBX and Asterisk servers. 2. Custom Bot Development: The bot should be responsible for calling, receiving, and monitoring calls directly from the software with features such as: - Automated call routing - Call recording - Call analytics - Call forwarding - Interactive Voice Response (IVR) - Real-time call monitoring Ideal skills would include a deep understanding of VOIP technologies, especiall...

...communication system. The goal is to implement key features such as call recording, call forwarding, and managing SIP trunk. We are open to using any open-source alternatives like FreePBX if it aligns with the project requirements. Responsibilities: Integrate Asterisk with Node.js for seamless communication. Implement call recording functionality. Set up call forwarding features. Manage SIP trunk configurations. Provide guidance and documentation for future maintenance. Requirements: Proven experience in integrating Asterisk with Node.js. - Familiarity with VoIP technologies. - Previous projects demonstrating call recording and call forwarding implementations. - Knowledge of SIP trunk management. - Strong communication and problem-solving skills. - Preferred Qual...

...ensuring a streamlined and clean installation while removing redundant files. Technical Requirements - Proficiency in WordPress and PHP Laravel, with a strong understanding of their integration and functionality. - Experience with database management, particularly with MySQL, and handling large datasets. - Capability in API integration and familiarity with communication technologies such as Twilio/WebRTC. - Demonstrated expertise in conducting platform migrations with minimal downtime and data integrity preservation. Timeline and Deliverables Aware of the typical duration such projects demand, we have allocated a strict timeline due to pressing business needs. The client website must be completed by February 7, 2024, and the CRM by February 11, 2024. This accelerated schedule n...

...and integration with MEDIASOUP for WebRTC connections. Key Requirements: --- Audio Playback: Implement seamless audio playback on a Raspberry Pi using ALSA or OpenAL. --Audio Streaming: Integrate MEDIASOUP library for WebSocket communication, enabling real-time audio streaming. --GPIO Interaction: Utilize Raspberry Pi GPIO pins for input (Transmit Button) and output (Audio Producer). Configure GPIO pins to trigger audio transmission upon button press. --SSH Configuration: Enable SSH access for dynamic configuration of MEDIASOUP "Room" parameter. Skills and Experience Needed: -Proficiency in C++ programming. -Experience with Raspberry Pi development. -Familiarity with audio processing (playback, streaming). -Prior experience with MEDIASOUP and WebRTC is hig...

...is key, and thus a fully integrated live chat feature is central to my vision. Key Requirements: - Robust streaming capabilities for multiple sports - Live chat functionality with emojis, GIFs support - Advanced moderation tools for a safe, inclusive chat environment - Secure user registration/login system for chat access Ideal Skills: - Experience in live video streaming technology (Webrtc, HLS) - Proficiency with real-time communication (WebSocket) - Knowledge in implementing user authentication systems - Strong grasp of frontend and backend development - Prior work with chat systems and moderation tools is a plus The platform should be user-friendly, scalable, and maintainable, with a focus on providing a seamless viewing and interactive experience. If you�...

...to connect and share live video content with high quality. Key Features: - Robust user profiles for a personalized experience - High-quality livestream capabilities - Scalable to support a growing user base Requirements: - Strong experience with social networking sites - Proven track record with livestreaming technologies - Exceptional coding skills in JavaScript/HTML5/CSS3 - Familiarity with WebRTC or similar streaming protocols - Experience with responsive web design for cross-device compatibility - Capable of integrating with various social media APIs Ideal Candidate: - Creative thinker with a portfolio showcasing similar projects - Attentive to the latest web trends, especially in social media - Excellent communication and time-management skills I'm excited to find ...

...glitches. - Audio Frequency & Tone Detection: Pick up specific tones and frequencies in the audio. - Utilize the header pins in and out of the raspberry PI. The audio client will integrate with a library called MediaSoup for connection, utilizing WebRTC. As such, prior experience with MediaSoup and WebRTC would be valuable. Skills and experience that are advantageous include: - Proficiency in C++. - Experience with Raspberry PI. - Understanding of audio processing (Playback, Streaming, Frequency, and Tone Detection). - Experience with MediaSoup and WebRTC. I believe this project provides an intriguing challenge for any individual armed with the right skill set. Your expertise will bring this project to life. PLEASE research Mediasoups library before bidding...

instale astpp basado en astpp, el problema que presento es que luego de haberlo instalado en debian 10 no me permite realizar conexiones sip, cabe destacar que desactive el firewalld y fail2ban para ver si era el problema y aun no me permite realizar conexiones

Looking for someone with expertise in nodejs along with prior experience in using webrtc to resolve some errors in my app

Sales :- • Conduct market research to understand the financial needs and preferences of target customers. • Prepare sales materials, presentations, and proposals to engage with prospective clients. • Source new clients and get repeat business from existing clients. • Sell Investment products I.e. Mutual Funds SIP & Lumpsum, Bonds, FD’s.

As part of the website testing strategy, I need a Python script that will parse a Git commit's change log. This script shall: 1.) Accept as a parameter a URI to a Git repo; 2.) Accept as a parameter a branch name; 3.) Accept as a parameter a commit hash; 4.) Produce as machine parsable output a set of files that were changed during the specified commit. 5.) Be written using Python 3.13 Any third-party libraries used to perform this task need to be approved prior to inclusion into the deliverable.

Actualizare magazin online adaugare produse, prelucrare fotografii, actualizare / adaugare descrieri, actualizare caracteristici, adaugare/modificare preturi. Alte activitati specifice, necesarei bunei desfasurari. Platforma integratoare este Easysales, aceasta platforma gestioneaza toate marketplace-urile partenere. Nevoia noastra este de a exporta toate produsele noastre pe mai multe marketplace uri. Avem nevoie de o persoana cu experiente similare proiectului nostru. Va multumesc si asteptam propunerile dumneavoastra.

I am looking for a Python expert, ideally with experience in VoIP, Call-center's Dialler softwares and SIP protocols. I need help in building a python application(just a script/server side python app) that will connect to VoIP/Dialler and get the audio stream for call and reply with an audio stream. I've a Dialler(MicroSip) setup on one of my servers, I have used the JS example and LinPhone application to make calls between 2 apps and it works fine. What i want to achieve is: Write a python program that should recieve, and make calls using computer's Michrophone and Speakers and should have the option to add anyother audio stream in reply to a call instead of getting the audio from system's michrophone. I've gone through plenty of libraries but as i am no...

...skilled and experienced freelance iOS Developer to assist in the implementation of a screen share feature within our existing framework. The ideal candidate should have expertise in WebRTC, SFU, and a strong background in iOS development. Responsibilities: - Collaborate with our team to understand the requirements for the iOS screen share feature. - Design, develop, and implement the screen share functionality on iOS devices. - Integrate WebRTC and SFU technologies to enable real-time screen sharing. Qualifications: - Proven experience as an iOS Developer with a focus on freelance projects. - Strong expertise in WebRTC and SFU technologies. - Previous experience implementing screen share features in iOS applications. - In-depth knowledge of iOS frameworks, desig...

I'm looking to develop an Android app for random video chatting that works similar to LiveTalk (). Upon loading, this app should provide two login options - "Login with Google" and "Login as Guest". Once the user is logged in, the primary function of the app needs to be to allow the user to sta...onward, it will call backend API to get common room id for consequent two users and connect them in a call. If any one party disconnects the call, it should automatically connect to the next person. This also needs to have admob ads integration. . Here are the skills and experience ideal for this job: • Experience in Android app development • Expertise in real-time video chat functionality using webrtc • Backend API development.

I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.

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