oleglab
diff --git a/‎sip-to-webrtc/README.md‎ renamed to ‎sip-over-websocket-to-webrtc/README.md‎
Lines changed: 6 additions & 3 deletions b/‎sip-to-webrtc/README.md‎ renamed to ‎sip-over-websocket-to-webrtc/README.md‎
Lines changed: 6 additions & 3 deletions
diff --git a/‎sip-to-webrtc/main.go‎ renamed to ‎sip-over-websocket-to-webrtc/main.go‎ b/‎sip-to-webrtc/main.go‎ renamed to ‎sip-over-websocket-to-webrtc/main.go‎
diff --git a/‎sip-to-webrtc/softphone/constants.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/constants.go‎ b/‎sip-to-webrtc/softphone/constants.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/constants.go‎
diff --git a/‎sip-to-webrtc/softphone/inboundcall.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/inboundcall.go‎ b/‎sip-to-webrtc/softphone/inboundcall.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/inboundcall.go‎
diff --git a/‎sip-to-webrtc/softphone/invite.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/invite.go‎ b/‎sip-to-webrtc/softphone/invite.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/invite.go‎
diff --git a/‎sip-to-webrtc/softphone/rcmessage.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/rcmessage.go‎ b/‎sip-to-webrtc/softphone/rcmessage.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/rcmessage.go‎
diff --git a/‎sip-to-webrtc/softphone/register.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/register.go‎ b/‎sip-to-webrtc/softphone/register.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/register.go‎
diff --git a/‎sip-to-webrtc/softphone/sipmessage.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/sipmessage.go‎ b/‎sip-to-webrtc/softphone/sipmessage.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/sipmessage.go‎
diff --git a/‎sip-to-webrtc/softphone/softphone.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/softphone.go‎ b/‎sip-to-webrtc/softphone/softphone.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/softphone.go‎
diff --git a/‎sip-to-webrtc/softphone/utils.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/utils.go‎ b/‎sip-to-webrtc/softphone/utils.go‎ renamed to ‎sip-over-websocket-to-webrtc/softphone/utils.go‎
@@ -1,6 +1,9 @@
-# sip-to-webrtc
-sip-to-webrtc demonstrates how you can connect to a SIP over WebRTC endpoint. This example connects to an extension
-and saves the audio to a ogg file.
+# sip-over-websocket-to-webrtc
+SIP Signaling via WebSocket is defined in [RFC 7118](https://www.rfc-editor.org/rfc/rfc7118.html).
+If you want to connect to a SIP server via UDP/TCP see [sip-to-webrtc](../sip-to-webrtc)
+
+sip-over-websocket-to-webrtc demonstrates how to connect to a SIP Server via Websocket.
+This example connects to an extension and saves the audio to a ogg file.
 
 ## Instructions
 ### Setup FreeSWITCH (or SIP over WebSocket Server)