Ideas / tasks for making Jingle audio calls great again
Documentation
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update https://dev.gajim.org/gajim/gajim/wikis/help/gajimfaq#general -
add wiki page about audio calls
Codecs
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Add iLBC and GSM as low bandwidth fallbacks for older voip clients (?) -
Add AMR-WB andAMR codecs. -
Figure out how to set the bit rate for Opus.
DSP
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Echo canceller
UI
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Info about the used codec, bit rate and STUN -
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
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STUN auto discovery.
Testing
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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
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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)?