XSF Communications Team - 2020-10-08

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

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.

Martin: thanks

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

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

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

Ok, will prepare, thanks

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?

Anything I forgot?

Sorry, should be "about September" I think ✏

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

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

Im fine with a color change!

Our qualified readers give the good hints! 😃

Tweet is out

Great!

https://twitter.com/xmpp/status/1314147875144839169?s=20

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

Uhh

😱

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.

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

as well as the signalisation around

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

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

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

i can ask, they're my colleages :p

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

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