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SUMMARY:Bridging WebRTC and SIP using GStreamer & SIPjs
DTSTART;VALUE=DATE-TIME:20230926T073000Z
DTEND;VALUE=DATE-TIME:20230926T080000Z
DTSTAMP;VALUE=DATE-TIME:20260411T215351Z
UID:indico-contribution-250@indico.freedesktop.org
DESCRIPTION:Speakers: Sanchayan Maity (asymptotic.io)\nWebRTC has become u
 biquitous as the technology powering various forms of video experiences on
 line. While Session Initiation Protocol (SIP) predated WebRTC by 12 years\
 , and had become the predominant protocol used to set up real-time media s
 essions between groups of users\, WebRTC looked to add real-time media i.e
 . audio\, video to every web browser without the need of a separate soft p
 hone client.\n\nWhile WebRTC has become the de-facto standard for real tim
 e communication on the Internet\, SIP still sees use in some scenarios\, s
 uch as bridging to phone networks (PSTN) and physical conferencing equipme
 nt.\n\nIn this talk\, we talk about how we went about connecting WebRTC an
 d SIP systems using GStreamer and SIP.js.\n\nhttps://indico.freedesktop.or
 g/event/5/contributions/250/
LOCATION:Palexco Room 2
URL:https://indico.freedesktop.org/event/5/contributions/250/
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