Certain enhancements, from my experience I wish I knew sooner (#760)

* Remove unused ErrTurnCredentials as it is no longer used. It was used when CredentialType was an option, but it is now deprecated. Ref: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#credentialtype

* Enhance WebRTC documentation with reliability and tests

Added details on configurable reliability and interoperability tests.

* Enhance clarity on WebRTC signaling process

Clarify the roles of codecs, media, and data channels in WebRTC signaling. Specify that TURN may be used for peers behind restrictive NAT, not always P2P.

* Add SFU and MCU explanations to WebRTC document

Added sections on Selective Forwarding Unit (SFU) and Multipoint Control Unit (MCU) to explain their roles in WebRTC. If someone develops a multi-user videoconferencing app, sooner or later they need to know this.

Author: ris-work

webrtc-explained.md

@@ -64,10 +64,10 @@ Protocol used by WebRTC Data Channels.
 
 Provides:
 - multiple streams
-- partial reliability
+- configurable reliability
 - no head-of-line blocking
 
-Runs inside DTLS.
+Runs inside DTLS. Can be configured to be TCP-like or UDP-like or anything in between.  
 
 ## DTLS (Datagram TLS)
 TLS over UDP.
@@ -78,6 +78,8 @@ Used for:
 - securing SCTP
 - securing SRTP
 
+DTLS does not typically go through CA Root Verification in WebRTC; it is verified in the signalling phase using the key fingerprint. 
+
 ## mDNS
 Masks local LAN IP addresses for privacy (e.g. *.local hostnames).
 
@@ -126,3 +128,13 @@ Commonly includes:
 - timers
 - UUID helpers
 
+
+## Interoperability 
+Check [WebRTC interoperability tests](https://github.com/sipsorcery/webrtc-interop) for samples in other WebRTC libraries like libwebrtc, pion, SipSorcery and whether they can interoperate with each other. 
+
+## Extras
+### Selective Forwarding Unit (SFU)
+This is not peer-to-peer WebRTC; a central server manages the media streams. This is how certain "mute" functions or noise detection is implemented in multiuser video conferencing. Used to restrict audio/video flow from users, due to administrative controls or due to being "idle" in a video/audio call.  
+
+### Multipoint Control Unit (MCU)
+This is an alternate architecture to the above, still not peer to peer but the central server itself takes the responsibility to mix/re-encode the streams.  

webrtc-signaling-explained.md

@@ -16,7 +16,7 @@ This includes exchanging:
 WebRTC peers cannot discover each other on their own.
 
 They need to:
-- agree on codecs and media setup
+- agree on codecs, media and datachannel setup
 - share network information
 - exchange security fingerprints
 
@@ -75,7 +75,7 @@ Contain:
 6. Peer B sends Answer back via signaling channel
 7. Both sides exchange ICE Candidates
 8. ICE finds a working route
-9. Peers connect directly
+9. Peers connect directly (or through a TURN relay, but still end-to-end encrypted)
 
 
 ## Signaling Server Responsibilities
@@ -95,7 +95,7 @@ It does *not*:
 
 Once the WebRTC connection is established:
 - the signaling server is no longer required
-- peers communicate directly (unless TURN is used)
+- peers communicate directly (unless TURN is used when the peers are behind restrictive NAT)
 
 
 ## Simplified Diagram