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WebRTC, what’s in it for me?

2.
MY BACKGROUND

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1
Pion
Collection of Go RTC libraries
- Building blocks to build your vision
- Designed to be flexible
- Community Owned/Non-Commercial
PeerConnection API in Pure Go
- CreateOffer/AddTrack/….
- User controls media pipeline
- SettingEngine
- Interceptors

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2
Pion
TURN as an API in Pure Go
- Callbacks for authentication
- Bring your own logger
- Run TURN and HTTPS on same port
- Embed TURN in application
Ion
- Cluster based system for building RTC
- SFU Instance
- AVP Instance (Processing real-time media)
- LIVE Instance (SIP/RTMP/HLS Endpoints)

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WebRTC for the Curious
Book on how WebRTC really works
- Not just about the public APIs!
- Deep dive on protocols
History of WebRTC
- Interviews with RFC authors
WebRTC in practice
- Debugging every piece
- Teach the sharp edges

6.
WHAT IS WEBRTC?

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4
Protocol for Browsers
E2E Secure Connection between Peers
- You decide the topology
Multiple Audio/Video Tracks
Binary Data
- Can be lossy
- Can be unordered
- Multiple distinct DataChannels

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And More!
- aiortc (Python)
- GStreamer’s webrtcbin (C)
- werift (Typescript)
- Pion (Golang)
- Shiguredo (Erlang)
- |pipe| (Java)
- rawrtc (C++)
- webrtc-rs (Rust)
- AWS WebRTC (C/Embedded)
- ?

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Bundling of existing protocols
You might already know these!
- (S)RTP
- DTLS
- ICE
- SDP
WebRTC in 4 steps
- Signaling
- Connecting
- Securing
- Communicating

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WEBRTC IN A RTMP WORLD

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Lower Latency
You can expect sub 500ms
- TCP/TURN may effect this
- You can make a choice of quality vs latency
You can get it much lower though!
- Google Stadia sees 125ms for keypress [0]
- playoutDelayHint API
[0] https://www.pcgamer.com/heres-how-stadias-input-lag-compares-to-native-pc-gaming/

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Secure By Default
DTLS and SRTP
- TLS over UDP
- Self-Signed, fingerprint via signaling
You can use existing gateways/proxies

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Flexible Networking
NAT Traversal
- Video never has to enter your network!
- Feels like ‘automatic port forwarding’
- Also can route through a ‘WebRTC Server’
Stay connected as you roam
- Switch networks with ICE Restarts
- Congestion Control in userspace

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Advanced Media Capabilities
Upload Multiple Quality Levels via Simulcast
- Stop running expensive transcoders!
Add/Remove tracks at anytime on either side
- You can mix/match codecs
Fast moving and widely adopted
- H265 and AV1 Support (not all Agents yet)

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Binary/Text Communication
Great for sending metadata
- Users in a room, chat etc…
E2E Secure
- No reason to upload this stuff to backend
Great for real-time
- Can be unordered/reliable
- Avoid HoL blocking

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It’s a standard
Widely Available
- C, C++, Go, Python, Rust, Android, iOS….
Remove subjectivity from incompatibilities
- Community is directed and resolving issues
It’s in the browser
It’s easy to bridge
- Uses existing protocols from telephony

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WEBRTC IN A SRT WORLD

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Agents negotiate in-band
Codec Negotiation is in the protocol
- You can declare I support X only
- Don’t have to put this logic in your codec
Track Negotiation is in the protocol
- I am willing to accept n tracks
- I am going to send you n tracks

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Re-uses existing protocols
Widely Available
- C, C++, Go, Python, Rust, Android, iOS….
It’s in the browser
It’s easy to bridge
- Uses existing protocols from telephony
- If you want to view in the browser….

20.
WEBRTC IN A RIST WORLD

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A Simpler WebRTC
Optional Security
Negotiation isn’t handled by the protocol
- Interoperation will need to be solved

22.
OPEN SOURCE SHOWCASE

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ns-remote
16

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kerberos.io
17

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CloudRetro.io
18

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Neko
github.com/nurdism/neko

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TelloGo

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ascii

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cloud-morph

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ssh-p2p

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Snowflake

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webwormhole.io
25

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s4y/space
26

34.
Project
Lightspeed
27

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github.com/pion
pion.ly/slack
twitter.com/_pion
sean@pion.ly