I want to make traceroute in the browser via flash. I asked about ICMP packets, but another alternative would be UDP packets if I could set the TTL. However, I don't see that option in flash.net.DatagramSocket.
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I am developing a client in Flash and using http req/resp to communicate with the server. For a while the session works fine and then the connection is terminated by the server. I did a wireshark sniff at the server and the last message which it sends is a RST packet. Also it happens only when I'm using IE and the server and client are in different domains. This does not happen in FireFox. I have been struggling to find a sol, till I found this thread. It's a bit old but I am not sure if this bit of info is important but I am connecting to the server via a gateway.
View 1 RepliesI have a program in c++ that sends a socket of 23723 bytes to a flash player in one "shot".The flash player sometimes receives two packets of size 17520 and 6203, and other times it receives a single packet of 23723.
EDIT: There does not seem to be a way to obtain the total number of bytes associated with the send data from flash. This would make it very difficult to build a loop to reconstruct the "broken" packet.I thought TCP was supposed to correctly recombine the packets and I am having difficulty recombining them myself.This is my receiving handler in flash:
var socket:Socket = new Socket();
socket.addEventListener(Event.CONNECT, socketConnectHandler);
socket.addEventListener(ProgressEvent.SOCKET_DATA, socketDataReceivedHandler);[code]...
will at times print 23723, and at other times, will print 17520 quickly followed by 6203. The total size are just examples and are subject to change for each send sockets. Also many sockets can be sent or received at the same time, so the split packets can be inter-mixed. In c++ I could determine what the total size of the sent data was, even if I did not receive the whole sent data yet.
How I can calculate packet loss from Flex.
View 1 RepliesHow I can calculate packet loss from Flex.
View 1 RepliesMy script is loading an external SWF. Everything runs smoothly during the load. But when the packet arrives, I noticed that something else is inside: insanity.
How to reproduce:
- Create two flas
- In the first one, write the following in the first (and unique) frame: //
- In the second one, write the code to load the first:
Code:
trace("1")
var loader:Loader = new Loader()
loader.load(new URLRequest("Untitled-1.swf"))
stop()
Run the first to produce the swf, then the second. The output will be an endless stream of "1"s, clearly showing that it's looping. But the loader wasn't even added to stage yet and it has no action to play the main movie! And now, for the craziness coup de grace: remove the "//" from the first fla and export it again. Bingo, no more endless loops.
So we're trying to come up with a way to automatically assess the "quality" of streams published to an FMS. So far, I've come down to 3 different statistics:
* Round-trip ping time to client (using the Client.ping() method and Client.getStats().ping_rtt.
* A home-grown jitter calculation that compares the deviation between wallclock and media time for an inbound stream, e.g., I get N+/-M seconds of media over a period of N seconds of wallclock time. How big is M on average? (uses Stream.time and Date.getTime()).
* How much packet loss are we getting?
For this third metric, I have the client sending up period reports containing tuples of <media_time, cum_bytes_sent>, where media_time is provided by NetStream.time and cum_bytes_sent is provided by NetStreamInfo.byteCount. I'm then comparing these samples to a similar set of samples obtained on the server via Stream.time and Client.getStats().bytes_in. Unfortunately, these two data sources do not seem to be measuring exactly the same thing. While they are close to each other, the received bandwidth (e.g., measured by the server) always seems to exceed the sent bandwidth (measured by the client) by a small amount (e.g., 0.5% to 2%) on a clean network.
The goal of this whole exercise is to enable to server to automatically detect when a client is having problems (either network or local CPU). Among other things, this will trigger us to tell the client to try sending a lower-bitrate stream.
So, my questions:
* Is there a method for me to reliably calculate (or obtain) a packet loss factor? With something like RTP, I could simply look for holes in sequence numbers, but I don't see any APIs that would give me that sort of information via FMS (our inbound streams are usually RTMFP).
* If there isn't a way for me to get a clean packet loss factor, is there another quality metric I should be looking at?
Does any one have a idea about how to do a packet loss test like in ping test from flash technologies (Flash or Flex)?
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