XSF Communications Team - 2020-10-08

  1. jcbrand

    I don't work in libstrophe, only strophe.js

  2. jcbrand

    I don't work on libstrophe, only strophe.js

  3. Martin

    peetah: > by the way, if someone here could give an explanation about what are 'global timed handlers' introduced in libstrophe 0.10.0, that would help a lot this french translation :) Maybe you'll ask pasis, he's usually in.the profanity muc.

  4. peetah

    Martin: thanks

  5. emus

    > I don't work in libstrophe, only strophe.js > I don't work on libstrophe, only strophe.js thats what I confused

  6. Martin

    And if he's not in there jubalh can probably give you contact details.

  7. jcbrand

    emus: If you have a tweet for me for the newsletter, I'll send it out

  8. emus

    Ok, will prepare, thanks

  9. emus

    jcbrand: > The XMPP Newsletter on September 2020 > Read about the progress on iOS clients, Gajim, Movim and many more! https://xmpp.org/2020/09/newsletter-09-september/ > #xmpp #jabber You can take the pictures from here: https://twitter.com/xmpp/status/1309030534572580864?s=20 Anything else?

  10. emus

    Anything I forgot?

  11. emus

    Thank you very much!

  12. emus

    Sorry, should be "about September" I think

  13. ldkjgoiwe

    the last 6 newsletter tweets used the green image version already, I request the blue version this time! 😀 https://twitter.com/xmpp/status/1194969787912331265

  14. Martin

    We have 4 colours so we should pick the color by month%4. 🙂

  15. emus

    Im fine with a color change!

  16. emus

    Our qualified readers give the good hints! 😃

  17. jcbrand

    Tweet is out

  18. emus


  19. emus


  20. vanitasvitae

    It is now possible to do audio calls between XMPP accounts and SIP accounts: https://twitter.com/MovimNetwork/status/1297570000790867974

  21. emus


  22. edhelas


  23. la|r|ma

    edhelas, is that peer-to-peer or via a proxy that terminates the WebRTC? Because my SIP client afaik doesn't speak WebRTC but only ICE-UDP with ZRTP encryption.

  24. edhelas

    nan it's p2p, basicallt the transport is converting jingle messages into SIP SDP messages

  25. edhelas

    as well as the signalisation around

  26. edhelas

    so you need your clients to have compatible audio/video codec

  27. la|r|ma

    sure, but webrtc transport and encryption are not directly compatible with plain ICE-UDP transport and ZRTP encryption

  28. la|r|ma

    but apparently linphone supports SRTP-DTLS so I guess they also do the webrtc transports part

  29. edhelas

    i can ask, they're my colleages :p

  30. la|r|ma

    I just guess that my SIP telephone or the SIP connector of a standard router won't support DTLS

  31. la|r|ma

    Yeah, unfortunately can't call my home phone from Conversations, just doesn't connect properly, I guess because of missing WebRTC support on the SIP side