gajim issueshttps://dev.gajim.org/gajim/gajim/-/issues2023-04-25T00:24:21Zhttps://dev.gajim.org/gajim/gajim/-/issues/8615Add setting to change client language in preferences2023-04-25T00:24:21ZAlejandro SosaAdd setting to change client language in preferencesI can't change client language in preferences, Ubuntu 16.04 LTSI can't change client language in preferences, Ubuntu 16.04 LTShttps://dev.gajim.org/gajim/gajim/-/issues/9144Add support for XEP-0352: Client State Indication2021-01-02T20:04:19ZAndrey GurskyAdd support for XEP-0352: Client State IndicationChanges of status: available/not available, composing/paused and so on can generate more traffic than comes from useful real messages. Allow users to decide whether they need them at the moment.
Per default it can be controlled automati...Changes of status: available/not available, composing/paused and so on can generate more traffic than comes from useful real messages. Allow users to decide whether they need them at the moment.
Per default it can be controlled automatically by idle event (proposed by @lovetox). Additionally it would be useful to be able to enable/disable it manually.https://dev.gajim.org/gajim/gajim/-/issues/9540Ideas / tasks for making Jingle audio calls great again2023-05-28T10:14:38ZOliIdeas / tasks for making Jingle audio calls great again### Documentation
* [ ] update https://dev.gajim.org/gajim/gajim/wikis/help/gajimfaq#general
* [ ] add wiki page about audio calls
### Codecs
* [x] ~~Add iLBC and GSM as low bandwidth fallbacks for older voip clients (?)~~
* [x] Add ~~A...### Documentation
* [ ] update https://dev.gajim.org/gajim/gajim/wikis/help/gajimfaq#general
* [ ] add wiki page about audio calls
### Codecs
* [x] ~~Add iLBC and GSM as low bandwidth fallbacks for older voip clients (?)~~
* [x] Add ~~AMR-WB and~~ AMR codecs.
* [ ] Figure out how to set the bit rate for Opus.
### DSP
* [x] Echo canceller
### UI
* [ ] Info about the used codec, bit rate and STUN
* [x] Some icons for calling / hang-up? 📞
* [ ] Preferences: setting for bit rate preferences / max bit rate. could be just a slider. codecs are disabled or reordered accordingly (don't forget rtp overhead).
* [ ] Preferences: `[x] Enable STUN auto discovery (recommended)`
### Connection
* [ ] STUN auto discovery.
### Testing
* [ ] Jitsi
* [ ] Some qxmpp client (telepathy-nonsense?)
* [ ] aTalk
* [ ] Monal
* [ ] Asterisk
* [ ] Sylkserver: Echo and Playback
* [ ] Jingle-SIP-Gateway. If that works, test SIP clients:
* [ ] Linphone
* [ ] Fritzbox
* [ ] native Android SIP client
* [ ] baresip
* [ ] PhonerLite
* [ ] MicroSIP
### Open questions
* Is it possible to use a TURN server with farstream?
* Are there any web clients that do support Jingle audio?
* Is ICE supported (XEP-0371)?