raul2r2 Posted November 20, 2010 Share Posted November 20, 2010 Hi... I was analyzing a bit the source code and data structures and I'm a bit surprised that there isn't anything similar to an RTP protocol implementation. Seeing voice packets data type it hasn't got any kind of timestamp... and for my little understanding of the thing, the server is not doing some kind of "remix" of all client voice streams in only one to redistribute to the clients and save bandwith... so in theory the server is doing some kind of multicasting resending all voice streams separately to all clients connected, so in theory, the clients are consuming a lot more download bandwith than upload one when them could be the same. Please correct me if I am wrong... I was thinking in implement something similar to RTP/RTCP for the server, remixing all voice streams in only one to resend to each client, but I'm not so clever guy to see if this was previously has been taken in consideration by the developers and discarded for whatever reason that I can't understand in this moment... for example, because it would has a big penalty in latency costs. What do you think?Regards, Link to comment Share on other sites More sharing options...
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