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rainliu merged 5 commits into from
Dec 4, 2025
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Certain enhancements, from my experience I wish I knew sooner #760

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80b141e
Remove unused ErrTurnCredentials as it is no longer used. It was used…
ris-work Dec 23, 2024
8db78c9
Merge branch 'webrtc-rs:master' into master
ris-work Dec 4, 2025
a9c914e
Enhance WebRTC documentation with reliability and tests
ris-work Dec 4, 2025
2c01868
Enhance clarity on WebRTC signaling process
ris-work Dec 4, 2025
a8becf1
Add SFU and MCU explanations to WebRTC document
ris-work Dec 4, 2025
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16 changes: 14 additions & 2 deletions webrtc-explained.md
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Expand Up @@ -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.
Expand All @@ -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).

Expand Down Expand Up @@ -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.
6 changes: 3 additions & 3 deletions webrtc-signaling-explained.md
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Expand Up @@ -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

Expand Down Expand Up @@ -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
Expand All @@ -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
Expand Down
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