WebRTC - Is it ready? 2013
4 likes•5,372 views
WebRTC is an open-source project enabling real-time communication in web browsers through JavaScript APIs, offering secure audio, video, and data channels. It involves complex techniques for peer-to-peer connectivity, supporting various codecs and built-in features for media management, including echo cancellation and noise reduction. While supported by browsers like Chrome and Firefox, concerns exist over codec patents and potential limitations in compatibility with certain platforms.
WebRTC - Is it ready? 2013
- 2. Web RTC is: ● A free, open project that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs to conduct SECURE communications. ● Features: ○ ○ ○ ○ Secure Secure Secure Secure Audio Channel (CD Quality) Video (Full HD) Screen Sharing Data Channel
- 3. Web RTC Block Diagram
- 5. Some Complexities Designed for P2P communications so includes intricate mechanisms to traverse NAT using STUN, ICE and a TURN server which is a media relay.
- 6. WebRTC JavaScript API ● MediaStream (getUserMedia) ○ Microphone (Audio) ○ Webcam (Video) ○ Desktop ??? ● RTCPeerConnection ○ Media Multiplexing ○ Authenticate against identity providers ● RTCDataChannel ○ SCTP over UDP https://www.simpl.info/index.html
- 7. Built-in Voice Engine Capabilities ● NetEQ ○ Dynamic jitter buffer management ○ Packet Loss Concealment (PLC) ● Acoustic Echo Canceler (AEC) ○ Removes, in real time, the acoustic echo resulting from the voice being played out coming into the active microphone. ● Noise Reduction (NR) ○ Removes certain types of background noise usually associated with VoIP. (Hiss, fan noise, etc...)
- 8. Supported Audio Codecs ● G.711 ● Opus - Free CD Quality codec (Even Stereo) ○ 20ms frames ○ Variable Bit Rate 8K to 64K bps
- 9. Built-In Video Engine Video Jitter Buffer ○ Dynamic Jitter Buffer Image Enhancement ○ Remove video noise VP8 Codec ○ Royalty free ○ Full HD 1080p encoding
- 10. WebRTC Connection Flow 1. Obtain Local Media (GetUserMedia) 2. Setup Peer Connection (RTCPeerConnection) 3. Attach Media or Data 4. Exchange Offer/Answer (RTCPeerConnection)
- 11. Supporting Protocols ● Application Layer: ○ HTTP, WebSocket, SRTP, SDP, ICE/STUN/TURN ● Transport Layer: ○ TLS (TCP), DTLS (UDP), SCTP (for data channel) ● Network Layer: ○ IPv4 & IPv6
- 12. Peer Connection Diagram Pure Web
- 13. Peer Connection Diagram Over Legacy Network
- 15. Signaling Proprietary ○ RESTful ○ BOSH (long polling) ○ WebSockets Open Standards ○ XMPP over BOSH/WebSockets (14 years) ○ SIP over WebSockets (17 years)
- 16. Standards It's a joined effort ● IETF ○ Focus on protocol and interoperability ● W3C ○ Hight level APIs and device control (mic, camera, network) ○ PeerConnection API proposal
- 17. Concerns ● Video Codec Patent: ○ Since March 2013 Nokia has asserted a patent claim against HTC and Google for the use of VP8 in Android in a German court ● Browser Support: ○ Chrome, Firefox, and Opera ○ Apple and Microsoft have not publicly shown interest in the support of WebRTC in IE and Safari ○ (Note: Temporary IE supported via Chrome Frame)
- 18. Demos Canvas Integration: ● Webcam Toy (GetUserMedia + Canvas) ● Audio Recorder (GetUserMedia + Canvas) Video Chat: ● talky (GetUserMedia + RTCPeerConnection) File Sharing ● Sharefest (GetUserMedia + RTCDataChannel)
- 19. More Demos Games: ● Facekat (GetUserMedia + JS head tracking) ● Cube Slam (???)
- 20. Tools Browser Tool ● chrome://webrtc-internals/ ● adapter.js (http://www.webrtc. org/interop)
- 21. Mobile App Vonage iPhone - https://itunes.apple. com/app/id491391564?mt=8 Vonage Android - https://play.google. com/store/apps/details?id=com.vonage. TimeToCall
- 22. FreeSWITCH and WebRTC ● SIP Over WebSockets ● Media over SRTP ● Opus supported Issues: ○ Inconsistent client support ○ DTMF issues (INFO) ○ No video transcoding is supported Note: Asterisks Already supported WebRTC (but only with G.711 codec)
- 23. Business Applications ● ● ● ● ● Call Center PAAS (Open Source API) WebRTC enabled SBC Video Conference Outsourcing Development Vendor
- 24. Call Center Applications Teledini - Display web visitor key information when call arrives. Receive call from browser.
- 26. Talkdesk
- 27. Platform as a Service Apidaze - API service for Voice, SMS, WebRTC and worldwide DID numbers. XML style structure. Crocodile RTC - Javascript SDK for Call, Video, IM, File Sharing, and Payment system over their platform. Requestec - Mobile SDK, virtual meeting with billing, recording, appointments, and collaboration.
- 28. Requestec
- 29. Hardware Solution Genband - WebRTC to VoIP gateway with RESTful API Ingate - VoIP enabled smart enterprise firewall now optimized also for WebRTC. Realtime traffic shaping Sansay - WebSBC, REST API, Proprietary signaling (not SIP)
- 30. Video Conference (PAAS) Bistri - Free video chat service that binds all your social network contacts.
- 32. UberConference
- 33. Opentok
- 34. Other PubNub - High performance and low latency communication of arbitrary data
- 35. Using WebRTC on Mobile Device ● Open Source VoIP engine (and Video) ○ ○ ○ ○ ○ ○ ○ ○ ○ ○ Active community support PLC - Packet Loss Concealment Jitter Buffer handling CD Quality Codec RTP at the core RTCP implemented for QoS monitoring SRTP for secure media transport NAT traversal by ICE ECHO Cancellation & Noise Reduction Mobile Integration contributed by Vonage Everything one might need to build their own Mobile Native Client
- 36. Presentations from WebRTC conf ● http://www.webrtcworld. com/conference/west/presentations.aspx
- 38. Thanks