From 6f0265621c7b8a8df02b138e25a363d0576c655c Mon Sep 17 00:00:00 2001 From: Sean DuBois Date: Mon, 18 Dec 2023 11:01:33 -0500 Subject: [PATCH] Rename sip-to-webrtc to sip-over-websocket-to-webrtc --- .../README.md | 9 ++++++--- {sip-to-webrtc => sip-over-websocket-to-webrtc}/main.go | 0 .../softphone/constants.go | 0 .../softphone/inboundcall.go | 0 .../softphone/invite.go | 0 .../softphone/rcmessage.go | 0 .../softphone/register.go | 0 .../softphone/sipmessage.go | 0 .../softphone/softphone.go | 0 .../softphone/utils.go | 0 10 files changed, 6 insertions(+), 3 deletions(-) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/README.md (67%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/main.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/constants.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/inboundcall.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/invite.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/rcmessage.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/register.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/sipmessage.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/softphone.go (100%) rename {sip-to-webrtc => sip-over-websocket-to-webrtc}/softphone/utils.go (100%) diff --git a/sip-to-webrtc/README.md b/sip-over-websocket-to-webrtc/README.md similarity index 67% rename from sip-to-webrtc/README.md rename to sip-over-websocket-to-webrtc/README.md index 8ab13d3..b67ef61 100644 --- a/sip-to-webrtc/README.md +++ b/sip-over-websocket-to-webrtc/README.md @@ -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) diff --git a/sip-to-webrtc/main.go b/sip-over-websocket-to-webrtc/main.go similarity index 100% rename from sip-to-webrtc/main.go rename to sip-over-websocket-to-webrtc/main.go diff --git a/sip-to-webrtc/softphone/constants.go b/sip-over-websocket-to-webrtc/softphone/constants.go similarity index 100% rename from sip-to-webrtc/softphone/constants.go rename to sip-over-websocket-to-webrtc/softphone/constants.go diff --git a/sip-to-webrtc/softphone/inboundcall.go b/sip-over-websocket-to-webrtc/softphone/inboundcall.go similarity index 100% rename from sip-to-webrtc/softphone/inboundcall.go rename to sip-over-websocket-to-webrtc/softphone/inboundcall.go diff --git a/sip-to-webrtc/softphone/invite.go b/sip-over-websocket-to-webrtc/softphone/invite.go similarity index 100% rename from sip-to-webrtc/softphone/invite.go rename to sip-over-websocket-to-webrtc/softphone/invite.go diff --git a/sip-to-webrtc/softphone/rcmessage.go b/sip-over-websocket-to-webrtc/softphone/rcmessage.go similarity index 100% rename from sip-to-webrtc/softphone/rcmessage.go rename to sip-over-websocket-to-webrtc/softphone/rcmessage.go diff --git a/sip-to-webrtc/softphone/register.go b/sip-over-websocket-to-webrtc/softphone/register.go similarity index 100% rename from sip-to-webrtc/softphone/register.go rename to sip-over-websocket-to-webrtc/softphone/register.go diff --git a/sip-to-webrtc/softphone/sipmessage.go b/sip-over-websocket-to-webrtc/softphone/sipmessage.go similarity index 100% rename from sip-to-webrtc/softphone/sipmessage.go rename to sip-over-websocket-to-webrtc/softphone/sipmessage.go diff --git a/sip-to-webrtc/softphone/softphone.go b/sip-over-websocket-to-webrtc/softphone/softphone.go similarity index 100% rename from sip-to-webrtc/softphone/softphone.go rename to sip-over-websocket-to-webrtc/softphone/softphone.go diff --git a/sip-to-webrtc/softphone/utils.go b/sip-over-websocket-to-webrtc/softphone/utils.go similarity index 100% rename from sip-to-webrtc/softphone/utils.go rename to sip-over-websocket-to-webrtc/softphone/utils.go