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incipit: back to work on Flash/AIR a couple of months ago after years, since 2011 - a lot of time. I'm a flash adept on client-side so, later this lot of time I was on papers, today I was sure to find a deeply tested and so well working live videoconferencing subsystem embedded in Flash/AIR with the new "winning" RTMFP+H264+P2P software capabilities of the more recent releases of the flash runtime.. but today, by my new recent point of view, later a month of tests, I found in real life it works more or less like Sorenson video subsystem (2004) if not worse -> in stream lightness. today it is 2015, everything should be already tested in deep. so, what's wrong? is it my implementation to introduce stream lag? :-]
guys, does anybody has seen 1 RTMFP+H264+P2P really well working app in the web? with 'working' I mean that videocommunications works like .. yes.. like skype, which is the king in videocommunication: perfect audio, fullsize video - which is not really required but works so good also on mobile, zero lag time. (btw hangouts is a joke.. it seems Sorenson in fullscreen:DD)
My app - the app that I use for my tests - is able to make a perfect videoconference NO-FULL-SCREEN with RTMFP+H264+P2P only in a local area network. If I test it over an adsl link with ONLY 1 collegue - with not a performing computer - the communication is simply crappy (in only 2!!!! in this case skype worked perfectly but ONLY with a not perfect video). no matter which audio/video configuration I choose (1/8/20 fps? 1%/73%/90% quality? h264 not h264? speex or nelly moser? etc): flash is not able to make _stable_ quality videoconferencing. btw I also scale streams fps on livedelay >0.5 but also livedelay seems not to correspond to the reality of the stream congestion status.
.. I'm in blue screen: is the Adobe work a joke in the real scene? Where is the magic of skype? or am I missing something? [to test my app just click on "join audio/video" button, link below]
anybody has a url to show me of a working RTMFP+P2P app? (320x240)
edit: I googled a looot. There is no working demo published on the web which is considerable seriously well working for me. Isn't it? (Let me see.) Is skype the only final winner?? :///
btw:
- I use Red5 Server for RTMP and I'm testing MonaServer for RTMFP/P2P
- I use a propietary server to manage the logic of the streams of a multiuser live conference so I do not implement NetGroups: i just get the peer id from the streaming/peering server to establish DIRECT_CONNECTIONS
Massimiliano Carli
Message was edited by: Massimiliano Carli
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.. obviously today it worked well ..
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.. I'm trying to get something near to true-realtime streams.. I mean lag zero streams like skype, which seems impossible, but in any case I'm trying to get the best results possible.
now I'm making some tests trying monitoring SRTT, droppedFrames, audioLossRate to scale down automatically (and re-scale up when possible) each single stream in case of cpu/bandwidth problems. Will it be enough?
in any case I notice that sometimes one stream or another seems to stop by itself without appreciable reasons / software catch. seems a problem. (any feedback?)
(I still don't say nothing about e.g. speex: in my experience I simply get a lot of echo without headphone with or without speex.. and it seems true in any occasion. I prefeare nelly)
..
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why flash live encoder subsystem is so lagged inside?!
10 points