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#webrtc

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Another question for my #WebRTC friends: Do you know what *exactly* makes an audio stream sync up with a video stream in libwebrtc?

Is it the a=msid:<stream-id> <track-id> grouping on the sender side?

Is it the a=msid:<stream-id> <track-id> grouping on the receiver side?

Is it both? Or is it something else entirely, e.g. do we need to create a MediaStream from both tracks on the sender or the receiver? Or do we need to attach both audio and video tracks to the same <video> element?

@steely_glint @lminiero @danjenkins @s

After taking a closer look at #XMPP clients for the #linux desktop, there's this frustrating "tie" for finding a favorite.

#Dino, which is likeable for being able to do video and audio calls, only has limited support for multi-user chat (complete with fancy moderation tools). These audio and video calls it can do are AMD64-only at this time.

#Gajim, which is likeable for being able to do multi-user chat well (with great moderation tools), alas, can't do audio and video calls to the other XMPP clients (like, say, #Conversations, as they use a newer #WebRTC-based method now).

So there's this strange situation where one is tempted to use both at the same time.

My takeaway is that #Conversations for #Android is the only xmpp client that I would possibly and carefully recommend to family and friends at this time, as it can cover all of the above. (#Monal on #iOS/#MacOS only has "partial" support for Multi-user chat, BTW.)

Yes I'm aware of the existence of Snikket, Quicksy, and Prav. No need to chime in on those.