From 2db85c41ccee6ce3772d1e70684660f3c00eda47 Mon Sep 17 00:00:00 2001 From: Nirbheek Chauhan Date: Sat, 28 Oct 2017 19:02:56 +0530 Subject: [PATCH] Protocol.md: Fix headings --- webrtc/signalling/Protocol.md | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/webrtc/signalling/Protocol.md b/webrtc/signalling/Protocol.md index 1af892eee3..18ef5fa430 100644 --- a/webrtc/signalling/Protocol.md +++ b/webrtc/signalling/Protocol.md @@ -1,26 +1,26 @@ # Terminology -### Client +## Client A GStreamer-based application -### Browser +## Browser A JS application that runs in the browser and uses built-in browser webrtc APIs -### Peer +## Peer Any webrtc-using application that can participate in a call -### Signalling server +## Signalling server Basic websockets server implemented in Python that manages the peers list and shovels data between peers -# Overview +### Overview This is a basic protocol for doing 1-1 audio+video calls between a gstreamer app and a JS app in a browser. -# Peer registration and calling +### Peer registration and calling Peers must register with the signalling server before a call can be initiated. The server connection should stay open as long as the peer is available or in a call. @@ -38,7 +38,7 @@ This protocol builds upon https://github.com/shanet/WebRTC-Example/ * Closure of the server connection means the call has ended; either because the other peer ended it or went away * To end the call, disconnect from the server. You may reconnect again whenever you wish. -# Negotiation +### Negotiation Once a call has been setup with the signalling server, the peers must negotiate SDP and ICE candidates with each other.