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jcbrand
I don't work in libstrophe, only strophe.js✎ -
jcbrand
I don't work on libstrophe, only strophe.js ✏
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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.
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peetah
Martin: thanks
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emus
> I don't work in libstrophe, only strophe.js > I don't work on libstrophe, only strophe.js thats what I confused
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Martin
And if he's not in there jubalh can probably give you contact details.
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jcbrand
emus: If you have a tweet for me for the newsletter, I'll send it out
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emus
Ok, will prepare, thanks
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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?
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emus
Anything I forgot?
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emus
Thank you very much!✎ -
emus
Sorry, should be "about September" I think ✏
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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
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Martin
We have 4 colours so we should pick the color by month%4. 🙂
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emus
Im fine with a color change!
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emus
Our qualified readers give the good hints! 😃
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jcbrand
Tweet is out
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emus
Great!
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emus
https://twitter.com/xmpp/status/1314147875144839169?s=20
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vanitasvitae
It is now possible to do audio calls between XMPP accounts and SIP accounts: https://twitter.com/MovimNetwork/status/1297570000790867974
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emus
Uhh
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edhelas
😱
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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.
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edhelas
nan it's p2p, basicallt the transport is converting jingle messages into SIP SDP messages
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edhelas
as well as the signalisation around
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edhelas
so you need your clients to have compatible audio/video codec
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la|r|ma
sure, but webrtc transport and encryption are not directly compatible with plain ICE-UDP transport and ZRTP encryption
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la|r|ma
but apparently linphone supports SRTP-DTLS so I guess they also do the webrtc transports part
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edhelas
i can ask, they're my colleages :p
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la|r|ma
I just guess that my SIP telephone or the SIP connector of a standard router won't support DTLS
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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