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2011 2017 W e b R T C 1 .
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WEBRTC, MOBILE CONSIDERATIONS AND VOICE OVER IP IETF e W3C 0 c - - PowerPoint PPT Presentation
WEBRTC, MOBILE CONSIDERATIONS AND VOICE OVER IP IETF e W3C 0 c - - PowerPoint PPT Presentation
WEBRTC, MOBILE CONSIDERATIONS AND VOICE OVER IP IETF e W3C 0 c . 1 r u Google C o T s R - Microsoft n b e e p Apple W o ... 2011 2017 WebRTC (Real-Time Communications) Acquiring audio and video
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WebRTC (Real-Time Communications)
- Acquiring audio and video
- Communicating audio and video
- Communicating arbitrary data
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WebRTC (Real-Time Communications)
- Acquiring audio and video
- Communicating audio and video
- Communicating arbitrary data
MediaStream (aka getUserMedia) RTCPeerConnection RTCDataChannel
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WebRTC (Real-Time Communications)
- Acquiring audio and video
- Communicating audio and video
- Communicating arbitrary data
MediaStream (aka getUserMedia) RTCPeerConnection RTCDataChannel
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MediaStream
MediaStream MediaStreamTrack: Video MediaStreamTrack: Audio Left channel Right channel Input Output
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MediaStream
MediaStream MediaStreamTrack: Video MediaStreamTrack: Audio Left channel Right channel Input Output
Constraints
- Media Type
- Resolution
- Frame rate
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RTCPeerConnection
- Signal processing
- Codec handling
- Peer-to-peer connection
- Security (Encryption)
- Bandwidth management
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RTCPeerConnection
- Signal processing
- Codec handling
- Peer-to-peer connection
- Security (Encryption)
- Bandwidth management
Media Peer Peer SRTP (Secure Real-Time Transport Protocol) DTLS (Datagram Transport Layer Security)
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Signalling
- Exchange Session Description Object
○ Codec to use ○ Security keys ○ Network information
- Any messaging mechanism (HTTPS, Websockets, XHR, ...)
- Any messaging protocol (SIP, XMTP, JSON, ...)
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RTCSessionDescription (SDP)
[OFFER] v=0
- =alice 2890844526 2890844526 IN IP4 host...
s= c=IN IP4 host.atlanta.example.com t=0 0 m=audio 49170 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 m=video 51372 RTP/AVP 31 32 a=rtpmap:31 H261/90000 a=rtpmap:32 MPV/90000 [ANSWER] v=0
- =bob 2808844564 2808844564 IN IP4 host…
s= c=IN IP4 host.biloxi.example.com t=0 0 m=audio 49174 RTP/AVP 0 a=rtpmap:0 PCMU/8000 m=video 49170 RTP/AVP 32 a=rtpmap:32 MPV/90000
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Media Signalling Signalling Peer Peer
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NAT (network address translation)
- Let multiple computers share the same IP address
- IPv4 address exhaustion
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Signalling Signalling Peer Peer NAT NAT
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STUN (session traversal utilities for NAT)
- What is my IP address?
- Simple server
- CHEAP
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Peer Peer Signalling Signalling NAT NAT STUN STUN Media
STUN
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TURN (traversal using relays around NAT)
- Cloud fallback if peer-to-peer fails
- Data sent through the server
- Ensure call works in almost any environments
- EXPENSIVE
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Signalling Signalling TURN TURN Media
TURN
Peer Peer NAT NAT
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Signalling Signalling
STUN + TURN
TURN TURN Media Peer Peer NAT NAT STUN STUN
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ICE (interactive connectivity establishment)
- Framework for connecting peers
- Find the best path for each call
- How?
○ Gathering candidates ■ IP address + port + transport protocol
- Directly attached network interface
- Server reflexive (STUN)
- Relayed address (TURN)
○ Connectivity checks ■ Sort the candidate pairs in priority order ■ Send checks on each pairs in priority order ■ Acknowledge checks received from the agent ○ Nominating Candidate Pairs and concluding ICE
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Alice Bob Media Cloud App IOs Android WebRTC SDP SDP Signalling Signalling
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Architecture: Small call
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Architecture: Medium call
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Architecture: Big call
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VoIP (Voice over Internet Protocol)
1940 2006 1970 1999 2018
PBX (Private Branch Exchange) SIP (Session Initiation Protocol)
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Asterisk Phone EBC SIP Client Voice VLAN VoIP Provider PSTN
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Asterisk Phone EBC SIP RTP Client Voice VLAN VoIP Provider PSTN Phone
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Phone Asterisk Phone EBC SIP RTP Client Voice VLAN VoIP Provider PSTN BGW 3 2 1
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WebRTC + VOIP
2006 2018 2011
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r c e 2017 W e b R T C 1 . 2016 J i v e w e b
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Asterisk EBC VoIP Provider PSTN Signaling Server HTTPS/WSS Browser SIP
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Asterisk EBC VoIP Provider PSTN Signaling Server HTTPS/WSS Media Server (Bob) SRTP Browser (Alice) TCP SIP RTP
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Asterisk EBC VoIP Provider PSTN Signaling Server HTTPS/WSS Media Server SRTP Browser TCP SIP Browser
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TCP Phone Asterisk EBC VoIP Provider PSTN BGW 3 Signaling Server HTTPS/WSS Media Server SRTP Browser SIP Phone SIP Client Voice VLAN RTP
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WebRTC + VOIP + Mobile
2018 2011
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PSTN Phone Oauth (PKCE) TURN Cell HTTPS/WSS SRTP TCP Signaling Server Media Server Outgoing call
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PSTN Phone Oauth (PKCE) Cell PushKit pushRegistry.register(with: [.voIP]) createSession(channelID) createChannel(pushToken, encryption) EllipticCurveKeyPair Notification Service Signaling Server Media Server Register on IOs SIP
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PSTN Phone Notification Service TURN Signaling Server Media Server Cell HTTPS/WSS SRTP TCP TCP APNs Incoming call Register to callkit Decrypt payload Encrypt payload voIP push Incoming call
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Pitfalls
- IPv6 mobile provider
○ TURN
- IOs13
○ VoIP push must register to callkit ○ DND must be server side
- Callkit
○ No customization
- Background
○ Callkit + ConnectionService
- Bandwidth + CPU
○ Frame rate ○ Resolution ○ Pause streams ○ Batch update participants
- Android
○ Audio Routes ○ Proximity sensors
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