Presentation at Astricon 2014 with Tim Panton about how to get the most of of WebRTC - making sure to deal with video bandwidth, asymetric calls and remembering that things are different to typical VoIP (SIP)
commercially - Quality matters Using Asterisk - Hybrid model with some PSTN Have Asymmetric calls - agents and users Are on a tight bandwidth and CPU budget Video not top priority (yet) Have a clue
N E T Agent Agent Agent Agent Agent DeskPhones WebRTC Audio HTML Asterisk App Server Agent WebRTC SIP Phones User PSTN Audio FAX SMS Twitter Chat Email Facebook Skype
- can penetrate NAT at cost of setup time DTLS - encrypted traffic but call setup costs CPU Web service - may expose your Asterisk SRTP - encrypted media costs some CPU too.
- high bandwidth/ low cpu 2 quality modes - landline or incomprehensible Opus - new comer - low bandwidth / high cpu flexible quality and error correction settings VP8 - free video codec - High but variable bandwidth H264 - licensed video codec - free if < 100k (IANAL)
video Opus None The browser knows best Int Agents ulaw None low loss on LAN + will end up on PSTN Ext Agent Opus High BW / low loss Can set minimum DSL standards for staff Ext User Opus Low BW / high loss Users share DSL with web/games etc Note - Digium does not support opus on Asterisk