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I am trying to create a website which implements webrtc functionality.But in all the samples and reference documents , I was unable to find how to select for a specific user to chat with. Do i need to have some server kind of setup to list all available users or anything like that. Any sort of ideas will be helpful.

For eg: In all the samples which i refered, the user was joining a particular room or a session and the other recipient also joins the same room for chat.But what i need is something similar to skype or hangout which shows the callee status before call, and want to get rid of this chat room concept.

Vipin YoYo
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There are three core areas of any real time communication service:

  1. Presence - identifying who's online and who's available to speak for a call. (e.g showing a contact list and identifying who's online)

  2. Signaling - initiating the call, exchanging IP addresses, negotiating capabilities, hanging up. In a lot of cases, Signaling and Presence can be combined into a single service.

  3. Media connectivity and streaming - actually getting "connected" to the other endpoint and streaming audio/video. It typically requires some assistance from the signaling service to get the initial set of local, stun, and turn addresses exchanged in order to establish a connection with the other node. It also includes the codec technology, streaming code, and rendering.

Unless something has changed, WebRTC, to the best of my understanding, is just about #3 - media connectivity and streaming. It still requires the website to provide signaling (exchanging the SDP messages).

There are some companies and open source projects that indicate they provide the complete solution, but I've never used any.

selbie
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  • i need some help on the first core area "PRESENCE". How i will check the status of the callee and list it to the caller. – Vipin YoYo Nov 15 '16 at 12:08