Build WebRTC Server with Standard and SIP Clients
$30-250 USD
Thanh toán khi bàn giao
I need a developer who can create a WebRTC server using Janus or FreeSWITCH.
We have a system that 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 ability to accept calls and then later perhaps have a client join via webrtc. This is all strictly on voice, there is no need for a frontend or video. There is some API integration that will have to be worked out with the codebase and with the help of our developers.
ID dự án: #32205083
Về dự án
5 freelancer chào giá trung bình$460 cho công việc này
Hi Dear, We will provide you Asterisk based complete solution and integrate with your APIs. First system will verify details by using APIs then play the IVR. We already developed the similir solution for our customer. Thêm
Yes I can Build WebRTC Server with Standard and SIP Clients, message me i am ready to start work from now.
Hello. Thanks for your job posting. I am full stack developer have deep knowledge in WebRTC real time communication and RTCPeerConnection its base. I have ever done file transfer using RTCPeerConnection and it is no pr Thêm
