The BaBL project Real-Time Closed-Captioning for WebRTC. Luis Villaseñor Muñoz 30 th April 2014
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1 The BaBL project Real-Time Closed-Captioning for WebRTC Luis Villaseñor Muñoz 30 th April
2 BaBL, version 1.0: Project Goal To develop a proof of concept WebRTC conference application that is able to use the WebRTC's data channel for transmitting real-time captioning. 2
3 BaBL, version 1.0: Final result 3
4 BaBL, version 2.0: Project Goal To develop a WebRTC multiconference application with some extra features based on speech recognition as realtime closed-captioning, instant translation or transcription storage. 4
5 BaBL, version 2.0: Milestones Multiconference WebRTC application Real-Time Closed-Captioning Instant translation Transcription storage 5
6 Multiconference WebRTC application WebRTC, what is it? WebRTC is a free, open project that enables web browsers with Real- Time Communications capabilities. Its goal: To enable rich, high quality, RTC applications to be developed in the browser via simple Javascript APIs and HTML5. [1] As stated in WebRTC.org. 6
7 WebRTC APIs MediaStream: For acquiring audio and video. RTCPeerConnection: For transmitting audio and video. RTCDataChannel: For transmitting data. 7
8 MediaStream navigator.getusermedia(constraints, successcallback, errorcallback); [2] Figure by Justin Uberti and Sam Dutton. 8
9 RTCPeerConnection Signaling: Session description, ICE, STUN, TURN Media engines: Codecs, echo cancelation, noise reduction, jitter buffering Security: HTTPS, SRTP, DTLS 9
10 WebRTC architecture [1] Figure from WebRTC.org. 10
11 Signaling server NodeJS: Web server and signaling server. Fully implemented using Javascript. Socket.io: NodeJS module that enables websockets between clients and server. 11
12 Calling: The establishment User A Server User B Download webpage (HTTP/HTTPS) getusermedia Create room (websocket) Download webpage (HTTP/HTTPS) PeerConnection New user joined (websocket) Offer (websocket) Join room (websocket) Offer (websocket) getusermedia PeerConnection createoffer createanswer Answer (websocket) Answer (websocket) ICE candidates (websocket) ICE candidates (websocket) Media streams (SRTP) 12
13 Calling: The mesh Room A Room B 13
14 Calling: ICE/STUN/TURN Interactive Connectivity Establishment (ICE): RFC Candidates for IP address. Session Traversal Utilities for NAT (STUN): Request and response. Traversal Using Relays around NAT (TURN): STUN extension. Relay. Useful but resource-intensive. 14
15 Multiconference accomplished! 15
16 Real-time closed-captioning Web speech API: SpeechRecognition interface: For converting the voice into text. WebRTC data channel: For sending the text to the other peers. 16
17 Web Speech API Another HTML5 API: Specification by W3C. Only implemented on Chrome: The voice is sent to Google s speech recognition web service. A JSON object with a list of possible matches is returned. They use it for voice searches: 17
18 RTCDataChannel Bidirectional peer to peer: Really low latency. Secure: Datagram Transport Layer Security. Unreliable or reliable: Latency or accuracy. 18
19 Challenges Subtitles should be switched on/off by the remote user We send the remote user s requests using the signaling server. Continuous recognition We keep a list of user requesting subtitles. Microphone permission We use HTTPS. 19
20 Architecture Signaling server 2. Subtitles request 1. Subtitles request 5. Subtitles User A User B Google server 4. Subtitles 3. Voice 20
21 Real-time closed-captioning accomplished! 21
22 Transcription storage Keeping record of our conversations: Text is much lighter than audio or video. And easier to find! Indexed DB: One more HTML5 API. Local storage in the client side. 22
23 Transcription storage accomplished! 23
24 Instant translation Translation services online: They are not free. Microsoft Translator API: Free 2 millions characters/month. 24
25 Challenges Should go through the server My private developer key can t be in the client side. When to request the translation? isfinal flag. Not so real-time. But much cheaper! 25
26 Architecture Microsoft Translator Server 7. Translated subtitles 6. Subtitles 8. Translated subtitles Signaling Server 2. Subtitles request 5. Subtitles 1. Subtitles request User A Google Server 4. Subtitles 3. Voice User B 26
27 Real-time translation accomplished! 27
28 Wait! Last minute addon! 28
29 Spoken translated subtitles Speech Synthesis API: The other interface included in the Web Speech API. Chrome has some built-in speech engines. 29
30 Spoken translated subtitles accomplished! 30
31 Conclusion Not perfect: Programmed by just one person. Using free resources. These technologies are still under development. A little more time, a little more resources: And Sci-Fi won t be Sci-Fi anymore! 31
32 References [1] Google Chrome team. WebRTC.org. [Online; accessed 30-April-2014] [2] Justin Uberti and Sam Dutton. WebRTC. [Online; accessed 30-April-2014] [3] J. Rosenberg. Interactive Connectivity Establishment (ICE). [Online; accessed 30-April-2014] 32
33 Questions? 33
34 Acknowledgements Don Monte and Nishant Agrawal Elias Yousef Javier Monte Condeoliva and Miguel Camacho Ruiz Tania Arenas de la Rubia Carol Davids 34
35 Thank you 35
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