I would like to use Flash's RTMFP peer protocol but I am wondering if its worth developing for a mainstream audience? From what I understand it uses UDP and unless firewalls/routers of the users are configured correctly it becomes useless.
View 4 Replies
Is there an open source implementation of rtmfp?Is there FMS hosting that support rtmfp?
View 2 RepliesI have been using streaming Flash video but just found out that it can't be seen in many places because Flash streams over ports 1935, 443, and 80, which are often closed by firewall. Progressive only needs port 443 so it's more likely that it will come through firewall.
View 4 RepliesI have a flash application which has alternative content when Flash is not available, this works fine. However if a firewall blocks SWF files the SEO content does not show. The flash plugin is still installed so the alt content is hidden but what remains is just a blank screen where the Flash object should be.
View 1 RepliesWill the use of HTTP Dynamic Streaming in place of the RTMP protocol aleviate the connectivity problems experienced by users who are sitting behind restrictive corporate firewalls?
View 2 RepliesOk so from all my research I'm fairly positive P2P is not possible in Silverlight 4 (in browser). Flash can do P2P using Stratus/RTMFP which I believe uses UDP hole punching and a Stratus server to initialize the connection. Is there any plans for Microsoft to add some kind of p2p rendezvous service in the future? Can we get a definitive yes/no on whether or not p2p is possible in SL4.
View 2 RepliesI am using Adobe Citrus and Air. The goal is to make a voice chat application with low latency. I take the sound samples that I get from the Microphone SAMPLE_DATA_EVENT encode them in using Native Process and speexenc.exe.
The question with method should I use for the sending of the samples. I tried object repication and the latency is to big.
What can BitTorrent do that RTMFP cannot? And what can RTMFP do that BitTorrent cannot?
View 1 Replieshost in my webpage an swf, and through Cirrus I could establish a connection between two different users, that worked like a charm, BUT, I had to pass through msn the peer ID in order to establish that connection.I know that this part must be resolved by a webservice.. but, how can I make one, and, more important, how does it work?
View 1 RepliesI see that the new Flash Media Servers were released. Looking over the information on the product pages, Flash Media Enterprise Server 4 is the only available product that supports RTMFP connections?
We currently have Flash Media Server 3.5 and our product utilizes RTMP connections, but I've been working with your Stratus Server and RTMFP and have found that the new protocol better fits our needs. However, the price for Flash Media Enterprise Server 4 is beyond our budget. Is there any possibility that RTMFP support are/will be available in the lower levels of Flash Media Server?
I need to create a simple video streaming without using Flash comunicator server or other streaming server.So I am thinking to use RTMFP but I haven' t found any good RTMFP tutorial for this issue.
Do you know tutorial for Video Streaming using RTMFP ?
Only the Flash Media Enterprise Server(FMES) has the "RTMFP peer introduction services" [URL] influxis created SmartMesh which enables p2p in Flash Media Interactive Server(FMiS). [URL] I couldn't find info on what does the peer introduction service do, what is the difference in capabilities of FMiS and FMES and what is the difference between RTMFP unicast and RMTFP multicast. Is it possible to create a p2p rtmfp connection between two clients with FMiS assuming I have a webservice that help clients exchange FarID?
View 1 RepliesI use RTMFP with NetGroup and Adobe Cirrus. Broadcasting settings are 320x240 with 5 fps.
However, in some cases it takes a long time for the webcam stream to appear (10-20 seconds, even if having a 1:1 connection). If the stream appears eventually, it looks like this:
I think this comes from a high keyframe interval and the subscribers needs to wait for the next keyframe, so that the stream appears properly. This happens in contrast to streaming servers, where the server caches the last keyframes from the publisher stream.
how I am able to tune the broadcasting, e.g. set the keyframe interval and other settings. Can you recommend a good setup? And, why is this issue not popular (I haven't found much information about it)?
I'm developing an online meeting system with audio/video sharing, using Adobe Flex 4 and Flash Media Server 4. I'm using the RTMFP protocol to make the transmission of audio/video which increases considerably the performance. The trouble now is that i must record the audio/video transmitted, but i figured out that when using the RTMFP protocol the FMS doesn't operate in the channels. So, how could i make FMS record this channels?
View 1 RepliesI would like to serve high quality video (H.264) between a server and a single Flash Player client with low latency (RTMFP). The single client will be controlling a robotic device and seeing the results through the video stream, so low latency is important.
Option A: The Flash Player supports low latency streaming of video between two P2P nodes via the RTMFP protocol in Flash Player 10. It uses the Spark codec to encode the video and as far as I know it is not possible to encode using any other codec in the player.
Option B: Flash Media Live Encoder does support live encoding using other codecs (On2 and H.264) but cannot act as an RTMFP peer - this option would require Flash Media Server in the middle.It seems I cannot stream video to a single client over RTMFP with anything other than Spark.
I have Flex application - it connects to FMS and joins to NetGroup named "default".So I have for example 4 clients connected to the same server in the same group. And looks like not all clients are connected to each other! Client1 sees Client2's streams and messages but doesn't see others. Same for Client3 and 4. I know UDP must not be blocked in networks in order to P2P connections work, and this can be a reason why not all members of group see each other.My question is - if there any ways to make all clients see each other? For example - if no P2P connection can be established - connect client via Flash Media Server or some proxy?
View 1 RepliesI've heard a couple of people say only 50% succeed, but that sounds suspiciously low and the people saying that are probably all repeating the same rumor. My own very limited tests with various friends & colleagues work 100% of the time, as long as they have Flash Player 10, but that's obviously not right either.
Do you have some data from a largish test or deployment of home/corporate users?
I have a swc plugin embedded in my product to stream live video to a group of recipients.The plugin uses the NetStream, NetGroup and NetConnection objects, the server I use to handle the RTMFP session is Influxis.One client creates a group as a publisher, and stream a live video.His Camera is enabled and I can see the video on the screen as well. The others connect to his group and play the video.Peer assisted Networking is enabled for both the publisher and the recipients.The plugin worked well in all 10.x versions (10.1/10.2/10.3)However, when upgrading to flash player 11.x, it stopped working.
-11.x to 11.x isn't working.
-10.x with 11.x isn't working.
-only 10.x to 10.x is working.
The server I use to handle the RTMFP protocol is Influxis.com I sent them the same question as well.The weirdest fact is that I'm using flash builder 4.6 with 4.6 sdk, which is aimed for flash player 11.1, and it is only working with 10.x flash players.
We are building a live RTMFP voice chat application with Cumulus. While the basic voice transfer works pretty easy using NetStreams, we have one big problem:There does not seem to be a way to manipulate the microphone data that the NetStream sends, and also not a way to manipulate the data the listening NetStream receives before it is played.However, this is exactly what we need. We do not want to transfer the normal microphone recorded audio, but first pitch it, then send it, then play it. Or first send it, then pitch it, then play it. But it seems that the whole audio recording, speex encoding, speex decoding and audio playback is completely encapsuled within the NetStream class.The only ways to achieve what we want (and all of them removing NetStream completely) seem to be:Send raw pitched audio data. That does work, but is of course a lot of data to send and will likely not work fast enough outside of our local LAN testing.
Pitch audio data, convert to ogg/mp3 using existing encoders for flash, send, decode ogg/mp3 and play. But this would mean encoding each and every sample packet that is received from the microphone, adding header stuff, etc. So this would likely not even yield that much of a benefit compared to raw audio data.2.1. This would actually be a good way if there was a Speex encoder/decoder for flash. But ironically, there is none other than the built in one (which is used for encoding/decoding audio in NetStreams) that cannot be explicitly used. ot offering it, Adobe...Send the data to the Cumulus server, pitch (and probably convert) there and send to the recipient. This would likely not even be that much faster than 1. and also throw away the exact benefit of RTMFP, P2P communication.
I built a simple rtmfp app using flash builder (I tested using the cirrus dev key to make sure it connects) but I cant seem to use the rtmfp service on my fms development server. Ive modified the index.html (in the webroot dir on the fms computer) to provide a link to a folder containing the html file and the swf file generated from my flash builder project. When I type in "rtmfp://localhost/" on my app I receive a "NetConnection.Connect.Failed" or a "NetConnection.Connect.Rejected" (if i do it directly on the fms computer).
Do I need to place my app in the "applications" directory to run properly? Ive checed the adaptor.xml file and everthing seems correct, I even added my public ip to the host port field, is there some confguration I missed?
Im not sure what Im doing wrong, Ive double checked my router and the fms computer's firewall thorugh a port scan and found the ports to be allowing traffic through.
I cannot find this anywhere, but is it possible to use RTMFP between two different SWFs?What I'd like to do is press a button on SWF-A and have a message appear on SWF-B. The two SWFs are completely different apps.I get that this is trivial if you are using a single SWF, let's say hosted on a web-page, but I need to do this with two stand-alone SWFs.If it is possible, what is the magic to hook them together? I was assuming it was to use the same GroupSpecifier and MulticastAddress, but that doesn't seem to be working.
View 6 RepliesI have established Flash Media Server 4.5 on the house computer with win7 a router wasn't present, the cord goes on a straight line to the computer, at me allocated IP. I want that it was possible to address to the server from the Internet. At first video was started only on a local computer, then after change resolved ип in "adaptor.xml" the starting page of the server has started to open with other computers and examples work, video plays. But to incorporate on "rtmfp" leaves only from that computer on which there is a server (under the test on http://cc.rtmfp.net/all green points and one more too with all green points) leaves that computers can incorporate to the server on "rtmp" as the example of the server with video translation works well, but on "rtmfp" connects only those at whom green points.
View 2 RepliesI was try to connect my FMS 4 development server (Redhat linux OS) in live which was not connected, but the same setup in local windows is connected.I do know whats the exact problem. Also i will stop the iptables in linux but no use.
View 6 RepliesI have an app that streams in video live, and archives it as it goes along.My question is this: What happens when I hit, or allow, 'pause' on a live stream, and then 'play'? Obviously the person on the other end streaming continues to do so, and my application continues to archive it, but what about the person receiving the stream?
Do they resume play from the archived copy, or do they seek ahead to the current point in the stream, missing everything they were paused for?
a client publishes a simple webcam and microphone. I noticed that if I use rtmfp the admin console shows between 30 to 200 messages dropped for a simnple webcam at 320x240 10fps quality 69 and audio speex at quality 6. sor after some minutes the video is unsynchronized with the audio. I turned off aggregateMsg and queue. if I use RTMP there is 0 dropped packets.
Is anyone can explain why FMS 4.0..1-r2009 on linux 64bits High Internet speed line on client side without any bandwidth problem last flash player
I'm writing a communication system, using Actionscript 3 and C#. Flash is used to communicate with RTMP server and stream video and audio using RTMFP p2p, but the actual client logic is handled in C#.
What I want to be able to do, is allow users to share files between each other also using a direct p2p connection. I have got the RTMFP connection working but I need to access the IP address of the remote user so I can initiate a connection to send or receive files.
I looked at the Adobe docs but I can't see anything except how to access the farID.
Is there a way to get the IP of a RTMFP stream?
I have a simple Flex RTMFP P2P video app in the same mold as the Adobe Cirrus VideoPhone Sample application. A problem I've been encountering in developing this app (the same problem occurs in the sample) is when you try to connect to a ghost Stratus instance i.e you try to call someone whose Stratus id is in the database but who is no longer on the page. So here's an example of what I mean:Let's say you go to the Adobe Stratus sample and connect as Dan. Then open up a new tab, go to the sample again and connect as Fred. If from this point, you (as Fred) call Dan everything will work fine. But, if you close the tab in which you connected as Dan, and then from the Fred tab try to connect to Dan the program will just hang.
I would have thought there would be a NetStream event that would be triggered if you tried to connect to a Stratus instance that is not longer online but I can't seem to find anything besides NetStream.Connect.Rejected which doesn't seem to be called.
the only information i found online is that RTMFP uses 128-bit cypher. Does anyone know what exact encryption technology is used?
View 1 RepliesI managed to run the FMS on an EC2 instance and I am able to use admin console. So I think I can say, that the server is running. Now I want to connect to the multicast application, which is delivered with the FMS as an example, via RTMFP. I can't get it to work... With an FMS 4.5 development server I didn't experience such problems. It seems to be a problem with the AWS. It might be, that I missed something to configure
View 1 RepliesI would like to get the NetStream width/height when receiving a RTMFP stream. This is important because the video component needs different measures when, for example, the user receives a 4:3 or a 16:9 stream.
Unfortunately, the onMetaData callback for NetStream does not work as it does for RTMP streams.
