Questions

Hi,I'm having a reliability problem with publishing live, h.264 streams from FFMPEG (using the rtmp publish ability now built into FFMPEG, where I can set an rtmp address as the output).After a random period of time (usually between an hour and several hours), the default "live" application in FMS 4 (or 3.5) eventually drops the connection from the FFMPEG encoder. This is what appears in the log for the application:"Dropping application (live/_definst_) message. Clients not allowed to broadcast message."After googling the error, I discovered that people have been reporting this error for several years, but there has not been a single response from any FMS experts or Adobe about what causes the error.Any thoughts on what is causing this error? If you need any more information about how to replicate the error, I'd be more than happy to comply.Thanks!-Tuan

Hey all,Newbies here.We have an application that so far uses AS3/AMFPHP, but we are finding this "request-response" model limited and feel we could make better use of sockets to deliver a more streamlined experience.The application is primarily a data-application containing no media streams.We want to open a socket to FMIS and use this as a pipe in which to stream data to and from the application. This is all good.Consider this situation however, when values can be adapted externally outside of the FMS application: Example:A user who connects with $0.00 at the start of a Flash Client session, may topup money, via paypal externally. At this successful point, the users database value has $10.00, but FMS is still $0.00.How can FMS be informed of an external event such as this and update the client accordingly in real time? Thanks,

I am trying to set up my video site to stream from the “fms server” and the instruction stated that I should create my own new folder in the application directory on the “fms server,” and I have created my directory, which I called “brigma_streams”, and I have copied files from {FMS-Install-Dir}/ sample/Application/vod to my new folder.I also configured the vod service in the FMS-Install-Dir/applications/(in my-New-Folder/Application.xml file). Here is the sample of my configuration below:<Streams>/;${BRIGMA_STREAMS_VOD_COMMON_DIR}</Streams><Streams>/;${BRIGMA_STREAMS_VOD_DIR}</Streams>And here is an example on how I modified to my XML streaming:src="rtmp://localhost/inetpub/wwwroot/brigma_streams/SanfordNSons768K_002.f4v"thumb="thumbs/SanfordNSons768K_002.jpg" />After all these configuration and modification it still didn’t work at all when I tested the video in streaming mode. Could some one please take a look at and point me to the right direction. Thanks

Ok I stream live to my server (FMS 4.5) using the DVR feature and for it to work properly in the stream name I have to put "livestream.f4v or mp4:livestream.f4v" and works fine so I dont have any problem here, my problem is when I try to use the feature for IOS Devices + DVR. For IOS Devices in the stream name I need "livestream?adbe-live-event=liveevent" and work and I can seek in the IPad, Iphone but not in the flash player so I want to know what is the best way to use this two features together?

I want to use RTMP based HTTP DVR functionality and HLS based IOS functionality, So I need to know how to MPP from my existing DVR app to the livepkgr app so that the stream being recorded at the DVR app can be used by the Flash and Stream MPP over to livepkgr app and can be used by the IOS HLS.

Hello,I have some problems about installing FMS 4 on a CentOS 6 x64.After installing, FMS doens't start properly. I don't see the videos on the web interface. In the log file master.00.log, i can see these errors messages2011-12-0608:38:461675(i)2581226Edge (1748) is no longer active.- 2011-12-0608:38:461675(w)2581255Edge (1748) _defaultRoot__edge1 experienced 1 failure!- 2011-12-0608:38:461675(i)2581224Edge (1798) started, arguments : -edgeports ":1935" -coreports "localhost:19350" -conf "/opt/adobe/fms/conf/Server.xml" -adaptor "_defaultRoot_" -name "_defaultRoot__edge1" -edgename "edge1".FMS listened on 1935 port.I deactivated the firewall and selinux, but it doesn't workI have before make an install on a CentOS 32bits, and it worked well.Thanks for your help,Pascal

In http streaming I can't see any activity in the Administration Console. is it normal?

We'd like to port our FMS app over to AWS, and FMS on AWS looks like the way to do that, but the only pricing I can find (here:  http://www.adobe.com/products/flashmediaserver/amazonwebservices/) is based on RTMFP connections.  Since we only use RTMP, how would our costs be determined?  Thanks.

I am trying to install Flash Media Server on IIS7.5  without Apache.  Are there any Whitepapers or instructions on how to do this?  We are a IIS shop and want to avoid using apache if possible.

I was wondering what the difference is between connecting to 1935 and 80.  Does one port have an advantage over the other?I know by default 1935 is blocked and it appears to be the perferred port to use, but does port 80 HTTP slow down the stream?

It seems possible to enter UV numbers in Adobe Connect without a "1" in front of the numbers, as this is the way FMG is configured for Adobe hosted. However, we are not able to configure our FMG server to get this to work. When dialing numbers without a "1", we get the error: "this number cannot be dialed at this time". If adding a "1" for the same number, everything works fine.   Would you happen to know if there is a configuration without FMG that allows this to happen, or is this strictly on the SIP provider side?   Thanks

Hi,I'm trying to set up the latest FMG/FMS to work with our video conferencing/media server.  Basic setup is fine, flash phone connects and calls are intitated.  G711u audio is good.  Video, both at a VGA and CIF resolution doesn't make it through the gateway.  A wireshark trace tells me a vidfeo RTP stream (h.264) is arriving.  But, nothing on the flashphone.Some pieces of the config are included here.  sip.xml:                 <Profile>                        <!-- Unique name for the profile. This ID is used in workflow to identify where the call will be placed -->                        <profileID> sipVideoPhone </profileID> 

Hello,we're running some FMIS 3.5.7 servers on which we are experiencing regular segmentation faults on high load. The stack trace from all of the crashes are identical:Program terminated with signal 11, Segmentation fault.(gdb) where#0  0x082158ed in FcPlayInfo::cacheStreamMediaContexts(FcPlayInfo const&) ()#1  0x082281b4 in FcRecordedStream::executeTransition(FcPlayInfo&, INetStreamPtr const&) ()#2  0x0822e4c7 in FcRecordedStream::executeBufferCmd(FcPlayInfo&, INetStreamPtr const&) ()#3  0x0822e6be in FcRecordedStream::execute(int, FcPlayInfo&, INetStreamPtr const&, int, bool) ()#4  0x082155c2 in FcPlayInfo::execute(int, INetStreamPtr const&, int, bool) ()#5  0x081ffba0 in FcNetStream::doBuffering() ()#6  0x0830ca04 in TCClient::onQueueEvent(TCObjectPtr&, void*) ()#7  0x081aee9f in QueueAPC::run() ()#8  0x0844420c in TCThreadPool::launchThreadRun(void*) ()#9  0x081be305 in FMSFiberPool::run() ()#10

Howdy All, I've been able to save the audio stream from a phone conversation with the following code application.stream = Stream.get("My Leg ID"); application.stream.record(); application.stream.play("My Incoming Stream Name");But the saved stream is FLV. Is it possible to save directly as an MP3 instead?Thanks!!

Is there an IDE or Editor for Server Side Scripting? I am currently using Notepad++ with formatting for Actionscript, but I would like something that can check the syntax before I deploy my application. I am losing hours hunting down missing parentheses. Thanks!!

Hello I just installed the authentication addin to my FMS 3.5 server (Windows 2008 Server - 32Bit.)  I created a user and have check that the user/password is correct; however every time I attempt to connect to the server via Flash Media Live Encoder 3.2 I receive user not found.  Again, I will go back to the utility on the FMS and check and the user/password combination is valid.  I'm stuck, I would appreciate any assistance.  Thanks.

GOOD NIGHT FOR ALLPLEASE HELP MEI HAVE A FLASH MEDIA SERVER AND IS WORK VERY VERY GOODI WANT TO MAKE USER NAME AND PASSWORD FOR MY CLIENTSLIKE THISrtmp://192.168.0.1/LIVE?USER=PASS/TEST I SEARCH IN THE INTERNET AND I CANT FOUND HELPBUT WHEN I ASK SOMEONE TELL ME YOU NEED PHPSO HOW CAN I USE IT WITH RTMPTHANKS VERY MUCH

Hi,People, can anybody give clear idea of call and accept method in P2P connection.Sureshkumar G

Hi. We are trying to stream live content to both flash and IOS devices using the FMIS 4.5 and the stock livepkgr application. We are using the FMLE 3.2, h.264, with the main concept AAC audio codec. We have set the streamsynchronization variable in the FMLE XML setup file to "true" Flash player equipped devices can see the stream fine, with no audio issues. For IOS devices, the audio is in sync when you first connect, but after about 30 seconds the audio is starting to get out of sync. If I stop the playback on the Ipad, and re-start, it is in sync again, but goes out again...Any ideas?

Hi, I'm pretty new to flash media server (and adobe technologies in general).I have an Air app that publishes a camera stream to the server, that can be played live by the same app on another computer/device.I would like to record to disk all audio/video passing through the server (on the server side) while streaming it at the same time. How can I accomplish this?Thanks for any help.

When using Adobe Flash Media Live Encoder, it doesn't seem to recognize my system speakers, so I can't record system audio. Any suggestions?

Hi,We want to develop a broadcasting application, there would be a main broadcaster on video, with a lot of chatters. There would also be some chatters going on webcam (6 max). I'm still checking out NetGroups, IP Multicast, RTMFP Multicast, Fusion etc. but I am still lost. I also want to make sure that people who can't support RTMFP can be a part of the program by using RTMP, if that's possible right now to combine RTMP/RTMFP in such a big broadcasting application.Thanks

Hi all,I have created a audio chat many to many chat using amazon hosting. The problem i am facing is that even with one to one audio chat, we are experincing a lag of like  2-3 seconds everytime a user speaks, which we believe is not good for audio chat. Can  anyone please guide us what should be done to optimize it.Any Help in this regard would be a great helpThanks in advanceRegardsSumeet Kumar

In my core.log file I'm receiving these error messages;Server detected a backward timestamp from 1973406 to 0 in file:C:\FMSHOME\applications\livepkgr\streams\_definst_\bbc1Server detected a backward timestamp from 1973406 to 0 in file:C:\FMSHOME\applications\livepkgr\streams\_definst_\bbc2Is this an issue caused by the encoder? I'm using a Viewcast Niagara to encode and not the FMLE. These are the only settings I have available. I'm using all default settings on the FMS 4.5 server.  . Thanks,Dave

Hello,We run about 20 FMS servers through Amazon AWS using the Adobe AMI: fms-4.0.1.8011-centos-5.5-ami-1 (ami-6a1deb03)On some of the servers we get fmscore segfaults, which kills the apps instances and make it impossible to launch new app instances. I am having a hard time finding out whether this could be due to errors in our code or if there is some kind of instability in the FMS. It does seems to happen only on the FMSs that are most heavily used.Below are extracts from various logs (message, core, master, edge). Can anyone see from those logs what might be wrong?Thanks in advance!var/log/messageJan  7 17:09:15 domU-12-31-39-18-08-75 Stream[30456]: Bad network data; terminating connection : chunkstream error:message bytes 128 are more than the specified message length 0Jan  7 17:09:15 domU-12-31-39-18-08-75 Adaptor[30456]: Bad network data; terminating connection : (Adaptor: _defaultRoot_, VHost: _defaultVHost_, IP: 70.252.9.75, App: V20111115/en_US_ComServ, Protocol: rt