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Smart SIP for webRTC

Smart SIP for webRTC. Dr. Alex Gouaillard CTO, temasys Communications Singapore. SIP – default. Have a sip address Login / register to your SIP server Place a call to another sip address. Need to know: extension, realm, proxy, ….. For both caller and callee. SIP – default.

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Smart SIP for webRTC

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  1. Smart SIP for webRTC Dr. Alex GouaillardCTO, temasys CommunicationsSingapore

  2. SIP – default • Have a sip address • Login / register to your SIP server • Place a call to another sip address. • Need to know: extension, realm, proxy, ….. For both caller and callee

  3. SIP – default SIP signaling over UDP/TCPSIP Registars Signaling Path Media Path Call Setup SIP S SIP S Register SIP Dev SIP Dev

  4. SIP – default • Have a sip address • Login / register to your SIP server • Place a call to another sip address. • Need to know: extension, realm, proxy, ….. For both caller and callee Call Setup SIP S SIP S Register SIP Dev SIP Dev

  5. SIP – KISS Call Setup SIP S SIP S Register SIP Dev SIP Dev

  6. Click to call – concept NO SIP ADDRESS NEEDED TO CALL • Scenario 1: Link or button (a clickable object) that once clicked start an (audio) call without requiring any other input or action from the user. • Scenario 2: Optionally, caller should be callable: if you try to call and it does not go through, the callee should be able to call you back.

  7. Click to call – webRTC+SIP - Base Pre-allocated SIP addresses.- does not scale- no load balancing HTTP Requests Signaling Path Media Path Call Setup 3 WS SIP S SIP S Register 2 3 1 Serves app with embedded SIP addresses for caller and callee 4 App SIP Dev JS implementationSIP stack

  8. Click to call – webRTC+SIP - Base Origin based(SIP server checks webapp name) HTTP Requests Signaling Path Media Path Call Setup 3 WS SIP S SIP S Register 2 3 1 Serves app with embedded SIP addresses for caller and callee 4 App SIP Dev JS implementationSIP stack

  9. Click to call – webRTC+SIP - Mismatches Connection: ICE, Transport: WS vs TCP UDP Ports management: BUNDLE and rtcp-mux, Encryption/Security: (DTLS-)SRTP Codecs Signaling Path Media Path SIP over WS SIP over UDP/TCP ICE Fixed IP Web RTC SIP Dev 4 ports 1 port, muxed VP8 H.264, H.263

  10. Click to call – webRTC+SIPTemasys Design HTTP Requests Signaling Path Media Path Modified:Origin and IP based auth.No SIP account needed. WS Mod. SIP S SIP S SIP/udp Fixed IP SIP/ws App WebRTC2SIPGateway SIP Dev ICE H.264, H.263, H.261 VP8, opus SIP in JS Temasys solution webRTC LEGACY

  11. Current Full DesignCall Set Up (1/3) REST/ HTTP Req. Signaling Path Media Path WS Mod. SIP S SIP S STUN WebRTC2SIPGateway SIP Dev App TURN Temasys solution webRTC LEGACY

  12. Current Full DesignCall Set Up (2/3) REST/ HTTP Req. Signaling Path Media Path Mod. SIP S SIP S STUN ICE ICE WebRTC2SIPGateway SIP Dev App ICE ICE ICE TURN Temasys solution webRTC LEGACY

  13. Current Full DesignCall Set Up (3/3) REST/ HTTP Req. Signaling Path Media Path Direct or TURN for mediaDepending on ICE Neg. Mod. SIP S SIP S WebRTC2SIPGateway SIP Dev App SIP TURN Temasys solution webRTC LEGACY

  14. Better Full DesignSeparate SIG and MEDIA paths REST/ HTTP Req. Signaling Path Media Path WS Mod. SIP S App SIP S SIP Dev WebRTC2SIPSig. Gateway MCU Temasys solution webRTC LEGACY

  15. Even Better Full DesignBookkeeping - dispatching REST/ HTTP Req. Signaling Path Media Path WS Mod. SIP S App SIP S SIP Dev WebRTC2SIPSig. Gateway MCU Temasys solution webRTC LEGACY

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