Webrtc sip uri työt

Hi. We are looking for someone who is proficient in customizing WordPress Plugins and PHP. WebRTC function should be added on my site. And the developer should be able to connect Zoom API to my site. Conversation would be in English, but fluent Japanese speakers are a big plus! Thanks.

Hello, I want a feature for my project. Imagine a voice call application. When I activate the speaker during the call, the microphone will not detect the sound coming from the speaker. The microphone will only detect the sound coming from the user. Note Microphone and speaker should be active at the same time. That is, the microphone should be actively listening while the spea...for my project. Imagine a voice call application. When I activate the speaker during the call, the microphone will not detect the sound coming from the speaker. The microphone will only detect the sound coming from the user. Note Microphone and speaker should be active at the same time. That is, the microphone should be actively listening while the speaker is outputting sound. Libraries like webrtc can ...

Currently I am using VOS3000 Latest Version, I need to be taught how to use it. My client will connect to the VOS3000 using IP and register. and the VOS3000 will set the Route for each client to the Operator's SIP CID. either use one CID or Random CID (Roundrobin, multiple SIP Operator Destination) and set each DID from the SIP Operator to each Client. and how to use a gateway that connects to several SIP providers. and how to use other features on vos3000. a brief overview as in this image.

I am looking for a freelancer who can help me...set up and configure my Twilio account. Here are the details of the project: Setup twilio account for using soft phones Simple IVR menue Sip trunk Preferred Communication Channel: - Both SMS and Voice Call Workflow: - I have a detailed plan that I want to implement with Twilio Technical Level: - I have an intermediate level of understanding with Twilio Skills and Experience Required: - Experience in setting up and configuring Twilio accounts - Knowledge of Twilio's SMS and Voice Call features - Ability to understand and implement a detailed plan - Strong communication skills to collaborate and finalize the workflow - experience setting up softphones/ sip routing If you have the necessary skills and experience, please a...

Hello, I want to work with WhatsApp, the solution should be a web page application based on sip+webrtc.1: User A make a call to whatsapp 2: Wahatsapp receive a text message 3: User A receive ringback 4: Whatsapp click on link and establish the video call are you able to do something like that Regards

WebRTC expert needed - EXPERTS ONLY Thank you

I have a WebRTC app browser extension that works, EXCEPT when the extension is closed. Written in react. We need the to maintain connections and make everything work whenever the extension is open OR Closed. 1st test is getting the incoming call notification to display when the extension is closed. Final -all features work when extension is closed. (Just like when the extension is open) We will share similar code written in jQuery that does work perfectly. Are you interested?

I am looking for a freelancer who can help me with the configuration of a Dinstar UC2000-VE device for SIP Trunking TO 3CX. Project Requirements: - Configuration of a Dinstar device for SIP Trunking - Integration with an existing PBX system - Configuration of a single Dinstar device Ideal Skills and Experience: - Experience with Dinstar configuration for SIP Trunking - Knowledge of PBX systems and their integration with Dinstar devices - Proficiency in network configuration and troubleshooting If you have the necessary skills and experience, please provide your proposal for this project.

...assist me with setting up and integrating SIP calls to WhatsApp call conversion. Requirements: - Familiarity with setting up SIP servers on Linux operating systems - Experience in integrating SIP calls with WhatsApp call conversion - Knowledge of the technical aspects involved in setting up and integrating SIP calls - Ability to troubleshoot and resolve any issues that may arise during the setup and integration process Skills and Experience: - Proficient in Linux operating systems - Strong understanding of SIP servers and client software - Experience in integrating SIP calls with messaging platforms such as WhatsApp - Excellent problem-solving and troubleshooting skills If you have the necessary skills and experience in setting up and inte...

We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX. Requirements: Inbound and Outbound Calling. Caller ID must be passed on incoming calls. Ability to have more than one Whats App number to work simultaneously

I am looking for a freelancer who can help me with a project that involves converting SIP voice protocol and RTP audio to WhatsApp voice call. I am open to suggestions and have no specific requirements in mind. I do not have a preference for the programming language, so the freelancer can use either Java or Python, whichever they are more comfortable with. The deadline for this project is flexible, although it would be preferable if it could be completed in a timely manner. Ideal skills and experience for this job include: - Experience with SIP voice protocol and RTP audio - Knowledge of WhatsApp voice call integration - Proficiency in Java or Python programming language

I am looking for a freelancer who can integrate SIP to Whatsapp call functionality for both mobile and desktop platforms. The integration should include basic call functionality without any advanced or customized features. Standard security features, such as encryption, should be implemented for this integration. i need use android as voip gateway. inbound calls via sip protocol into outbound calls via WhatsApp Ideal Skills and Experience: - Experience in integrating SIP to Whatsapp call functionality - Proficiency in mobile and desktop application development - Strong understanding of security protocols and standards

...if it's possible to use use a direct connection to communicate with the players. If it's not possible to use a direct connection (due to NAT or firewall), connect a TURN server instead. Once the connection is made, then send messages between some clients and then terminate the app. This is just the basic summary. Please refer to the PDF attached for the full specification. I don't want to use WebRTC due to its high latency, but instead need to use raw UDP packets (SOCK_DGRAM). This is both portable and performant. There are various NAT traversal libraries out there, but I prefer using PJSIP because Android and iOS devices are supported. If you prefer to use another library, please consult me first. Thank you CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION I...

Project Title: Webrtc Screen Sharing Application Description: I am looking for a developer to fix a Webrtc application with a specific focus on screen sharing functionality. The application should be compatible with web platforms only. Specific Requirements: - As I am not sure about the specific requirements for the screen sharing feature, I would appreciate advice from the developer on the best approach and implementation. In particular ice candidates. Ideal Skills and Experience: - Strong expertise in Webrtc technology and protocols - Experience in developing screen sharing functionality within a Webrtc application - Proficiency in web development languages such as JavaScript, HTML, and CSS - Knowledge of WebRTC APIs and libraries - Ability to provid...

Cautam un consultant GDPR din Romania care să ne ajute să verificam, să ajustez si sa realizam toate documentatiile necesare (Termeni și Condiții, Politici de Confidențialitate, Politici pentru Cookie-uri...) specificațiile software-ului meu. Proiectul implică revizuirea specificațiilor software finalizate și asigurarea conformității cu GDPR. Cerințe specifice: Cautam o revizuire generală a specificațiilor pentru a asigura conformitatea cu GDPR Crearea tuturor documentelor pentru conformitatea cu GDPR: Termeni și Condiții, Politici de Confidențialitate, Politici pentru Cookie-uri... Complexitatea software-ului este medie Abilități și experiență ideale: Experiență dovedită ca consultant GDPR sau într-un rol similar Cunoștințe solide despre regulamentele și cerințele G...

...new at using iOS and WebRTC. So, I need some help getting the settings right before I can work on this. If someone is very experienced in this and would like to work on the whole project, please provide quotes for the following milestones: 1.) Creating a working webRTC Adaptation in Swift of the heygen Java Script demo app. 2) Integrating the livestream into an ARKit project so the it can be used as an overlay onto the real word. Onto a rectangular book cover but also as a an alpha keyed free standing person (the author stands next to the book seemingly on the table without any vidoframe around him…) 3) S2T, T2S (but this might already be part of the open ai API, no?) 4) Simple U Expected outcome: full guidance and code on how to set up my system and run my fir...

I have a freshly installed "FreeSWITCH", and I want to configure it with the below example: The "Outbound Traffic" - Accept the SIP traffic from the Local "A" () SIP Server. - Redirect the Signaling to the Local "B" () SIP Server. - If the Signaling response is accepted, the traffic (Signaling and Media) will be redirected to the Local "C" () SIP Server. The "Inbound Traffic" - Accept the SIP traffic from the Local "C" () SIP Server. - Redirect the Signaling to the Local "B" () SIP Server. - If the Signaling response is accepted, the traffic (Signaling and Media) will be redirected to the Local "A" () SIP Server. All the traffic mentioned will ...

...Technical Implementation: • Create an admin dashboard using React or similar frontend frameworks. • Utilize backend APIs for admin functionalities (user management, dispute resolution). • Implement role-based access control for admin privileges. 8. Collaboration Tools: Technical Implementation: • Integrate third-party APIs or develop in-house collaboration tools (whiteboard, screen sharing). • Use WebRTC or similar technologies for real-time communication. 9. Escrow with Milestone Automation: Technical Implementation: • Develop logic for defining project milestones. • Automate milestone payments based on predefined completion criteria. • Implement error handling for failed milestone completions. 10. Niche-specific Communities: Technical Imp...

I am looking for a logo for my photo booth and mobile bar business. My business name is Sip & Snap. The logo should be fun and playful, with no specific colors or themes in mind. I am open to any suggestions. Skills and experience needed: - Graphic design - Logo design - Creativity - Ability to incorporate specific text and symbols

...communication between my players. Basically what I want is a C++ program that: Connects to a STUN server (probably my own STUN server using CoTurn) If it's possible to use a direct connection to communicate with the players, use that. Otherwise, connect to a list of backup TURN servers. Once the connection is made, then send messages between some clients and then terminate the app. I'd rather not use WebRTC due to its high latency, but instead prefer using raw UDP packets if possible (such as sendto, recvfrom etc). Is this something that can be done in C++? I notice that CoTurn has a library you can use, but there are also other ones such as Pjsip. CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH...

Looking SBC Ribbon Expert Communication between SBC and MS teams - call manager the issue shows codec issue -- i have change it but still same issue.. any one ? I will not pay until the issue resolved ------------ SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/TCP X.X.X.X:5060;branch=z9hG4bK-UX-ac19-5a05-0033 From: "XXXXXXX" <sip:XXXXXX@X.X.X.X:5060;user=phone>;tag=ac195a05-2a;sgid=2 To: <sip:XXXXX@X.X.X.X:5060;user=phone>;tag=44188389 Date: Mon, 27 Nov 2023 13:42:17 GMT Call-ID: call-BD0D1C00-0000-0010-100B-D@ CSeq: 2 INVITE Allow-Events: presence Server: Cisco-CUCM11.0 Session-ID: 00000000000000000000000000000000;remote=00000000000000000000000000000000 Content-Length: 0

We are looking for a skilled developer to create a WebRTC app for our Asterix Cloud PBX system integrated with Pipedrive. Preferred Platform: - The app should be developed for the web browser platform. Specific Features: - We have a detailed list of features that we require for the app. But basic functions like incoming call, outgoing call, integration for click to dial , log call history are the minimum Timeline: - We have a timeline for the project, but it is flexible. Ideal Skills and Experience: - Proficiency in WebRTC development - Strong knowledge of PBX systems and integration with Pipedrive - Experience in developing web applications - Attention to detail and ability to follow a detailed feature list

Create an HTTP-triggered Azure Function that does the following: - Get the JSON request body. Write values to two separate tables in Dataverse. - Vehicle events - 1 header line per function call - Wheel events - many rows per function call (~...called from known devices. - You can pass on secret keys, etc. Either way, you should be safe. - Authenticate to Dataverse using a managed identity Set up an Azure DevOps deployment pipeline and code repo for future maintenance and deployment. The request body contains base64 image data. This can be transferred to a Dataverse table or ideally stored in Azure Blob storage with only the Azure Blob URI stored in the Dataverse table next to the transaction. When the function completes, it responds to the endpoint with a "success" o...

Require experienced developer to build WebRTC browser based call app with suitable frontend and backend technologies. DO NOT MIND TO APPLY FOR THIS JOB IF THIS IS YOUR FIRST PROJECT RELATED TO WEBRTC or Voip Applicants must show previous projects or will not be considered.

hello, i have a mikrotik rb4011 router where i have a wan from isp on ether1 and a sip connect (without internet) on ether2. i need a config and correct routing on ether3 for pbx and config on ether4 for intern network and internet. you also can choose which router. maybe ccr2004 is the better one.

I am looking for a developer who can help me resolve an OAuth Google Drive Picker redirect error in JavaScript. Requirements: - Proficient in JavaScript - Experience with OAuth and Google Drive API - Familiarity with resolving Invalid Redirect URI errors - Ability to troubleshoot and debug OAuth configurations - Strong communication skills to discuss recent changes made to the OAuth configuration This project requires someone who can quickly identify and fix the issue with the redirect URI. If you have experience with OAuth and Google Drive API, please reach out to me.

I am looking for a VoIP developer who can create a VoIP application for my project. The ideal candidate should have experience in developing VoIP applications with the following functionalities: - Call routing and forwarding - Call recording and monitoring - Voice recognition and transcription I have no specific requirements or preferences for the programming language to be used, so the developer can choose the most suitable language for the project. The VoIP application is expected to be used by: - Less than 100 users - 100-500 users - More than 500 users If you have experience in VoIP development and can fulfill the above requirements, please submit your proposal.

...cyber panel cyberpanel control .................................................. .......... 3/ vm ubuntu server and config ip, and install nexcloud with domain name, complete and secure configuration, and add smpt messaging service and with IA config AND SSL, French language add. ............ STREAMING LIVE CREATE VM UBUNTU LTS 22 .04 4/ vm; creation of a vps for defusion of live video (webrtc or other ????Nginx + RTMP?????, from a phone or computer or other, I ask you which solutions are the best. demerge from one or more of my websites nextcloud I would like to decide for myself how much space I allocate for each client (check that it is possible to change it myself) for example add 1 TB OR 4 OR 6 OR 3 ........................................... !!!!!!! check that the s...

I am looking for a developer who can help me make a connection between my client and a specific SIP server using the PJSIP module and React Native. Preferred Skills and Experience: - Proficiency in working with the PJSIP module and integrating it with React Native - Experience in setting up and configuring SIP servers - Strong understanding of networking protocols and security measures - Familiarity with voice call functionalities and implementing them in mobile applications Project Requirements: - Connect the client's application to a specific SIP server of their choice - Implement basic security measures for the connection - Develop the application to support voice calls functionality If you have the necessary skills and experience, please reach out to dis...

...Technical Implementation: • Create an admin dashboard using React or similar frontend frameworks. • Utilize backend APIs for admin functionalities (user management, dispute resolution). • Implement role-based access control for admin privileges. 8. Collaboration Tools: Technical Implementation: • Integrate third-party APIs or develop in-house collaboration tools (whiteboard, screen sharing). • Use WebRTC or similar technologies for real-time communication. 9. Escrow with Milestone Automation: Technical Implementation: • Develop logic for defining project milestones. • Automate milestone payments based on predefined completion criteria. • Implement error handling for failed milestone completions. 10. Niche-specific Communities: Technical Imp...

I have a website for online chatting. Now I want to add a karaoke program. I already have webrtc members program that can play and listen to karaoke. But now I want to upgrade karaoke to a new level. I want to hire someone to make a Nodejs JavaScript program that I can iframe it to my chat website. 1. When the chat user presses button (Raise Hand to Sing), they will be put in a special list, this list is the person who pressed the button (Raise Hand to Sing), first will be on top, and the next person, and will be printed in : 1. nickname1, , 3. nickname3 ... This list will display for everyone to see. The person who press button (Raise Hand to sing) can lower the singing right to the back but cannot move up to front. For example, nickname1 can lower 1 person and it will be Nickna...

I am looking for a Linphone SIP expert to help me set up new features for social media app. I already have a Linphone SIP account set up and I want to implement additional functionalities like group chat and video conference Ideal skills and experience for this job include: - Strong knowledge of Linphone SIP and its features - Experience in setting up and configuring Linphone SIP accounts - Proficiency in developing and integrating additional functionalities into existing applications Should have hands on experience in setting up linphone video conference..

I am looking for a WebRTC developer to help me integrate audio/video calls into my existing application. The timeline for this project is less than one month. I am looking for a developer with expertise in making audio/video calls, screen sharing, and data channel communication. The developer should have experience with delivering to deadlines and working with existing applications. If you have prior experience with WebRTC development and have the skill set needed for this project, I would love to hear from you. Thank you.

I'm looking for experienced artist to help my medium-sized team (11-50 people) with a paint & sip activity at our office on December 1, 2023 from 12:30 PM to 3:00 PM to promote holiday fun and the work that we do as advocates for the working class. As it is the holiday season, the activities should be designed to raise the team's spirits and create a festive atmosphere.

I am looking for a freelancer who can help me with a project that involves using FREEPBX and asterisk ari to place outbound calls out of a sip trunk using pjsip. The purpose of the outbound calls is for customer service. I do not have any existing infrastructure to support this project, so it needs to be built from scratch. For project updates, I prefer communication through email. Skills and Experience Required: - Strong knowledge and experience with FREEPBX, asterisk ari, and pjsip - Previous experience with setting up outbound calls and sip trunks - Excellent problem-solving skills - Ability to work independently and meet project deadlines

...Technical Implementation: • Create an admin dashboard using React or similar frontend frameworks. • Utilize backend APIs for admin functionalities (user management, dispute resolution). • Implement role-based access control for admin privileges. 8. Collaboration Tools: Technical Implementation: • Integrate third-party APIs or develop in-house collaboration tools (whiteboard, screen sharing). • Use WebRTC or similar technologies for real-time communication. 9. Escrow with Milestone Automation: Technical Implementation: • Develop logic for defining project milestones. • Automate milestone payments based on predefined completion criteria. • Implement error handling for failed milestone completions. 10. Niche-specific Communities: Technical Imp...

Project Description: Sip Trunk user with PBX support I am looking for a skilled professional who can help me with the remote installation of a PBX system. The ideal candidate should have experience working with Asterisk PBX. Requirements: - Familiarity with Asterisk PBX system - Ability to remotely install and configure the PBX system - Proficiency in setting up new features for the PBX system - Experience in configuring voicemail services - Knowledge of call forwarding and interactive voice response (IVR) setup Skills and Experience: - Previous experience in installing and configuring PBX systems - Strong knowledge of Asterisk PBX system - Ability to troubleshoot and resolve any issues that may arise during the installation process - Proficiency in setting up voicemail services...

I am looking for a developer to implement GStreamer WebRTC on Jetson Nano for streaming media. The project requires working with an existing codebase and adding new functionalities. Ideal skills and experience: - Proficiency in GStreamer and WebRTC - Experience with Jetson Nano - Knowledge of streaming media protocols - Ability to work with existing codebase and add new functionalities I have a Jetson Orin Nano and 3 USB cameras. I need to have the Jetson nano attach the 3 camera sources to a webrtc stream and have a local signaling server. I then need to be able to connect to the webrtc stream with another computer browser located locally on the same LAN. I need a method to detect the signaling credentials from the client browser (I can help with this par...

Hello, I'm in search of help designing a EPS / SIP sandwich panel to be used in the manufacturing of one piece inground swimming pools. The panel would likely use polystyrene foam with steel or aluminum structural members and a metal skin and/or polypropylene skin. I have experience in the pool industry and do not need assistance with any of the hydraulic requirements, structural/mechanical engineering help only.

...looking to enhance the user experience by adding video and audio communication capabilities. Responsibilities: - Integrate video and audio call functionality using WebRTC. - Implement activity feeds to keep users updated on relevant events within the platform. - Collaborate with the team to troubleshoot and resolve any issues related to real-time communication features. - Ensure a responsive and user-friendly experience across various devices. Requirements: - Proven experience in Full Stack Development with a focus on Django and React. - Expertise in integrating video and audio call features into web applications. - Familiarity with WebRTC or similar technologies for real-time communication. - Strong knowledge of Django, Django REST framework, and React. - Experience wit...

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...Details about the eligibility criteria for the product. BankingProductFee: Information about any fees associated with the product. BankingProductLendingRateV2: Details about lending rates, including type, rate, comparison rate, frequency of calculation and application, interest payment due, repayment type, loan purpose, tiers, additional value, additional information, and additional information URI. Special Consideration for Rates Since some products have varying rates or comparison rates, the spreadsheet should be formatted to accommodate these unique situations. This may involve creating multiple rows for a single product or additional columns to capture different rates. Comparison Rate Interest Rate UPFRONT FEES ONGOING FEES DISCHARGE FEE EXTRA REPAYMENTS REDRAW FACILITY OFF...

i want to implement a simple architecture. at this moment my asterisk server works fine with endpoints and it can initiate calls in both directions. now i want to use a proxy server in between endpoints and my server . so im using dsiprouter with domain name of and its installed successfully. now there is a problem. i connect my sip users to dsiprouter. and dsiprouter to asterisk as pass thru. when i call any number with sip endpoint there is no problem and the call reaches the dinstar gateway and dinstar call the sim number(ip2tel). but the opposite side of call when sim user call the gateway and gateway route the call through asterisk. when asterisk send request to dsiprouter the request cannot find the endpoint i tried (realm-outbound proxy and etc) but i didnt make it

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I want a small project writen in .net blazor with a SIP Server integration. Server will have sip address username and password. Basic Ui for calling and answering. For demo purpose you can use any SIP server you want. The project will use 3CX SIP server. I suggest to use SIPSocery as library but feel free to use anything you want.

...recorded data will contain sensitive customer information, we require a high level of security to protect the data from unauthorized access. The project should be written in C# asp.net core 8. Video chat should start in the most online web environment using webrtc and a 3rd service (.net core API) should record this conversation. A link should be created for the meeting to start. When the participants click on this link, they should log in to the meeting directly. Postgresql should be used as the database. No other library other than webrtc should be used. There should be comments in code blocks Skills and Experience: - Experience with Web RTC technology and recording sessions - Strong knowledge of web development languages such as HTML, CSS, and JavaScript - Famili...

Project Description: I am looking for a skilled developer who can create a WhatsApp to SIP Gateway using Asterisk. The main function of this project is to enable call forwarding between WhatsApp and SIP. I require the gateway to be built on an open-source platform. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the destination's WhatsApp number. Skills and Experience Needed: - Strong experience with Asterisk and VoIP systems - Proficiency in working with both WhatsApp and SIP protocols - Knowledge of call forwarding and routing techniques - Familiarity with open-source platforms for building gateways - Ability to troubleshoot and debug any issues that may arise during the develop...

I am looking for an expert freelancer to integrate SIP services with mediasoup for my project. The purpose of this integration is to enhance the video calling capabilities on my application. Requirements for the project: - Experience: The freelancer should have expert level experience in SIP video integration with mediasoup This Project may require expertise in VoIP technologies along with WebRTC, Asterisk, kamailio etc. - Tools and Technologies: NodeJS is preferred, however to get the project integration we are open to suggestions and do not have any specific tools or technologies in mind. If you have the required expertise and can provide guidance on the best tools and technologies to use, please bid on this project.

Looking for someone experienced with freeswitch/lua and valet call parking. We need to send a sip_ph_P-Asserted-Identity header but it needs to match the caller ID for the call (keep in mind, inbound and outbound calls need to be differentiated). I already have a small but buggy "solution" that needs to be tweaked. We keep the CID info in freeswitch hash database and collect the data when the parked call is retreived.

I need to install a2billing + freepbx on my server. Create a SIP account and configure a trunk. It must be proven that it works. within 24 hours.