Gajim - 2019-02-07


  1. a45b Hello, was there an issue with the gajim.org XMPP server(s) yesterday?
  2. wurstsalat Server software update
  3. a45b Ah, I see. I had trouble connecting and wasn't sure if it was a client issue
  4. bot Carlos S created an issue in _gajim_ < https://dev.gajim.org/gajim/gajim/issues/9572 >: #9572: < Error retrieving history from openfire 4.3.2 with monitoring service 1.7.0 >
  5. bot Philipp Hörist closed an issue in _gajim_ < https://dev.gajim.org/gajim/gajim/issues/9572 >: #9572: < Error retrieving history from openfire 4.3.2 with monitoring service 1.7.0 >
  6. tom Does anyone how how I can get video and audio calling working in Gajim 0.16.X on Debian?
  7. tom or uh, Devuan to me more exact
  8. tom I would appreciate it
  9. asterix First, you should upgrade to latest 1.0.2
  10. asterix 1.1.2 sorry
  11. wurstsalat tom, with 1.1.2 you might be lucky to get audio calls working. Video does not work at the moment afaik
  12. tom wurstsalat, thank you, but I asked for help with 0.16.X
  13. asterix we don't support old versions
  14. wurstsalat Support for audio/video is broken in 0.16.x
  15. tom wurstsalat, yeah, I'm asking how to fix it
  16. tom It works fine on Gentoo, but Debian has issues
  17. asterix we can't say more that read the code and understand what's going wrong. It was never fixed because in 0.16 we depend on very old libs that are no more in distributions
  18. tom asterix, I understand if _you_ do not want to help me and that's your decision, I am asking anyone in general, not you specificly
  19. asterix ok sorry for having replied then, I won't anymore
  20. wurstsalat Guess it would've been fixed by now if anybody knew how to do that exactly
  21. tom asterix, oh no worries about that. I just didn't see it right to speak on behalf of everyone else here using 'we'
  22. wurstsalat Currently, priorities are others. Though, audio support received some love recently by oli
  23. tom wurstsalat, hmm. Well if you explain the problem I could probably fix it myself
  24. tom I am a Python programmer as well
  25. asterix I speak as Gajim developers (Lovetox and me), but sure other users are probably more available to tell you that dependences are no more in distributions.
  26. tom As I understand it it depends on libfarstream
  27. tom Is there a reason libfarstream was chosen over just piping things into ffmpeg?
  28. asterix because it does way more that that. It simplify code. Non need to re-invente the wheel each time
  29. tom like what?
  30. oli tom: i haven't looked into the code, but maybe you can get rid of farstream and use gstreamer directly
  31. tom Most likely I'll have to get rid of farstream, but I probably would not be replacing it with gstreamer. I really do not like gstreamer. I much prefer ffmpeg
  32. oli Most functionality is in gstreamer
  33. tom i see
  34. oli Farstream uses gstreamer. I don't know ffmpeg very well, but whats wrong with gstreamer?
  35. tom It's just not as high quality software
  36. tom I've dealt with both
  37. lovetox tom, the p2p connection, the audio protocol and codec negotiations, the detection and use of different audio output methods on different systems, thats all implemented in farstream and gstreamer
  38. lovetox if say you get rid of this all, you have nothing and start from scratch, i think you can do much better things with your time, just my opinion
  39. lovetox gajim only supplies the jingle negotiations which comes down to 3-4 stanzas sent over the xml stream
  40. bot Selen created an issue in _gajim-plugins_ < https://dev.gajim.org/gajim/gajim-plugins/issues/386 >: #386: < problem with pgp plugin in linux >
  41. ilac It looks like the ipv6 goes down from time to time for *.gajim.org.
  42. ilac Probably because of scaleway's usage of dhcp-pd.
  43. debacle oli, tom, lovetox, are you aware of aiortc?
  44. lovetox no, but aio stuff cant be used with gajim
  45. lovetox because it depends on using the python mainloop
  46. lovetox and we use the gtk mainloop
  47. lovetox and gstreamer has also a webrtc plugin
  48. lovetox https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/gst/webrtc-sendrecv.py
  49. lovetox although that is also a asyncio example
  50. lovetox but im sure we could use this also with the gtk main loop
  51. oli Gstreamer has great echo cancelation.
  52. oli Not sure if it can be used with farstream.