Asterisk webrtc integration for audio, video and screens share

We need some additional features in the telephony part of our application, for which we are looking at engaging someone.

Currently we have a b2b SAAS application thats used by our customers for sales and support management. The user can set up their customized IVR flow, do voice broadcast through our application. A new feature that we are adding is the click to call feature where the user's agent can click his customer's no. and a small pop up opens, now using webrtc, we connect the agent to the called no. While on the call, currently the agent has the option to mute, hold, unhold, hangup and transfer the call to another no. However, its on testing and still to be made live. We are using sipml for this and it has some issue with the unhold feature too that needs to be fixed

Now, we require a few telephony functionalities like group conferencing , transfer to an app flow and enhancements in our current implementation .

We can roughly divide these functionalities into 3 parts:

A. Webrtc Audio - Work required

1. Establish the webrtc currently implemented in view of scalability, robustness and voice quality and its integration with our application (we'll do the .net part but need help in the asterisk part of it). For this, as of no, we are just using one extension and there's no way to know who is making the call, so changes in asterisk context may be needed, to capture and keep the agent id and the no to dial from . Also, group audio conferencing and transferring the call to an app flow will be required

2. Configure webrtc for incoming calls too

3. Invoke the call pop up from the ivr and connect the caller to an agent

4. Enable parallel ringing when the connect to agent part of the flow in the ivr is arrived at.

B. Webrtc Video and screen share

1. Extend webrtc functionality in our web application for video calling, conferencing and screen sharing while making sure that the solution is robust and voice and video quality is achieved.

Since there could be many variables and options of technologies to be used to ensure best performance, this part can be further elaborated when work is actually started

C. Webrtc functionality in Mobile app

1. We are in process of creating mobile apps of our application. Once webrtc audio/video and screen share is integrated in the web application, we would want the same to be replicated in mobile app

If you can take up the project, pl provide your time and cost estimation for the same

Skills: Asterisk PBX, VoIP, PHP, Software Architecture, Mobile App Development

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About the Employer:
( 0 reviews ) Los Angeles, United States

Project ID: #21922460

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