From SIP to WebRTC and vice versa
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The document discusses the transition from SIP (Session Initiation Protocol) to WebRTC (Web Real-Time Communication), focusing on various media features and frameworks. It highlights the complexities involved in routing and the gateway functionalities of different servers like Sylkserver and Janus for facilitating audio/video calls and chat presence. Additionally, it mentions resources for troubleshooting SIP/WebRTC issues.
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From SIP to WebRTC and vice versa
- 1. SIP WebRTC From SIP to WebRTC Saúl Ibarra Corretgé | @saghul versa and vice
- 3. AG Projects
- 5. Audio / Video Chat Presence SIP beyond VoIP
- 6. WebRTC
- 10. MESSY SDP It’s a
- 12. CHANGE
- 13. From a TRAPEZOID Server 1 Alice Bob Server 2
- 18. VoIP Web + WebRTC Routing user@domain URL HTTP(S) Features (soft / hard) phone Browser Complexity ???
- 20. GATEWAY
- 22. Janus
- 24. SylkServer
- 26. SylkRTC API SIP XMPP SylkServer Core JanusSIP SIMPLE SDK
- 27. SIP audio/video calls SIMPLE Chat PresenceSylkServer WebRTC gateway Janus Presence and chat gateway
- 30. SylkServer 3 Audio / Video calls No transcoding Zero-configuration
- 32. N ext
- 35. In SIP / WebRTC trouble? bettercallsaghul.com