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| Technology scout | Topic | Rationale | Result |
|---|---|---|---|
| ICE, STUN and TURN: How WebRTC deals with firewalls and NAT | Addressing NAT and firewall traversal with peer-to-peer WebRTC communication | Investigation into important infrastructure component that could be offered by NRENs | Report: test report.pdf Demo: Screencast demoing PoC PoC service URL: https://brain.lab.vvc.niif.hu Source code of different components on github: |
| RENdez-Vous: One year of operational experience | Describes experiences of the French nationally deployed WebRTC desktop videoconferencing service | Harvest experience from first national deployment of a native WebRTC desktop videoconferencing service | |
| Distributed RENdez-Vous: An investigation into scale-out Jitsi | Assess possibilities for Jitsi software to scale out to a distributed deployment supporting all of EU R&E | RENdez-Vous would be a possible candidate for a shared EU R&E desktop videoconference system, if the choice is build-your-own. | |
| WebTut | A proof of concept implementation of an online web tutoring service | Possible lower integration cost for in-context communication, possible infrastructure component in NREN infrastructure | FCCN deployment: https://webtut.fccn.pt UNINETT test deployment: https://webrtc.uninett.no/webtut/home Source code of different components on github: https://gitlab.fccn.pt/sa8-webrtc/webtut-frontend/tree/develop |
| Service proposal for a Media API Service | Proposes a service offering high-level, real-time communication building blocks for contextual communication | Possible lower integration cost for in-context communication, possible infrastructure component in NREN infrastructure | |
| Screencast: Stream and record lectures with WebRTC | Investigate recording with WebRTC | Opportunity for simplification of lecture/screencast recording and streaming, new services possible | |
| Unified Communication and WebRTC | Assessment of impact of WebRTC on Unified Communication in R&E | R&E trend is towards UC deployments at every R&E institution | |
| SIP2webrtc gateway | Explore opening the legacy world of SIP for browser-based communication | Address interaction of new technology with installed base |
Useful links and information
Accessing WebRTC debug information in browsers
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Chrome: chrome://webrtc-internals
Debug FireFox
Firefox:
export NSPR_LOG_FILE=/home/ehugg/tmp/nspr.log
export NSPR_LOG_MODULES=signaling:5,mtransport:5,timestamp:1
export R_LOG_LEVEL=9
export R_LOG_DESTINATION=stderr
ICE media log:
For ICE/STUN/TURN:
Set R_LOG_DESTINATION=stderr
Set R_LOG_LEVEL=3 (can be anything between 1 and 9)
Set R_LOG_VERBOSE=1 if you want to include the module name
generating the message
For "signaling" (SDP offer/answer handling) and media transport, we use
the normal Mozilla logging infrastructure, which uses a comma-separated
list of modules, each one with its indicated log level; for WebRTC,
you'll be most interested in these:
Set NSPR_LOG_MODULES=signaling:5,mtransport:5
message to include timestamps.
Debug Chrome
google-chrome --enable-logging=stderr --v=4 --vmodule=*libjingle/*=9 --vmodule=*media/*=9linux log file:
.config/chromium/chrome_debug.logBasic info: https://www.chromium.org/for-testers/enable-logging
a) --vmodule=*source*/talk/*=3
b) --vmodule=*third_party/libjingle/*=3
c) --vmodule=*libjingle/source/talk/*=3
--enable-logging=stderr --log-level=3 --vmodule=*libjingle/*=3,*=0