Questions

Hi Adrian,We have same requirment, something like this.1. FMLE is streaming to AMS1.2. Need to feed same FMLE stream to AMS2, AMS3.3. finall we will have same streaming getting pulled/pushed to all AMS in my cluster.Can you help me to do this? seems this was your requirment.you are pulling stream from  Akamai and feeding same to server where you are running you NetConnect().Thanks, SJ

Is it possible to pull in an external rtmp link and use it as live source. Instread of using FME as the live source, I would like to re-stream another RTMP link.I've done this with ease in another rtmp streaming server, it's been while since I've done something like this using FMS/AMS.Thank you in advance.

Hi,Can someone please tell me where can I download Adobe media gateway developer version to run with Adobe media server 5 (developer version). I have already installed the Adobe media server 5 developer version.The best I found was this link "http://blogs.adobe.com/labs/archives/tag/flash-media-gateway" But there are no links on this page that lead to downloads.Please help.Regards,Sahil.

Hi to all,I have an AMS 5.0.3 that streams vod videos using both RTMP for Flash-enabled browsers and HLS for iOS devices.All RTMP accesses are logged in the access.xx.log files and there I can find all connect, disconnect, pause and seek requests and so it's possible to analyze the logs for finding out what portions of a video the users were watching.Now, I'd like to do the same thing for the HLS accesses. Unfortunately they are not logged in the access.xx.log files or in any other log files of AMS.The only entries I was able to find are in the Apache access logs. They look like this:93.135.58.250 - - [16/Feb/2014:18:55:01 +0100] "GET /hls-vod/183147.mp4.m3u8 HTTP/1.1" 304 - "-" "Apple Mac OS X v10.6.8 CoreMedia v1.0.0.10K549"93.135.58.250 - - [16/Feb/2014:18:55:01 +0100] "GET /hls-vod/183147.mp4.m3u8 HTTP/1.1" 200 8096 "-" "Apple Mac OS X v10.6.8 CoreMedia v1.0.0.10K549"93.135.58.250 - - [16/Feb/2014:18:55:01 +0100] "GET /hls-vod/183147.mp4Frag1Num0.ts HTTP/1.1" 200 1654776 "-" "Apple

Hi My app allows users to record video and audio via their webcams.  If the user stops speaking for more than a second or two there seems to be an interruption to the recording....so when they start speaking again the first second (or so) of their audio doesn't get captured.  It's almost like Flash falls asleep when there's no voice input and then wakes up slightly too late to start recording again.Ordinarily this wouldn't be an issue but my users want their recordings to be faultless and so aren't prepared to use the service unless every syllable gets recorded.Any ideas on how to fix this would be greatly appreciated!Elvin

I am using FMS 4.5 to record a live webcam stream. Because of the poor quality when recording both video and audio on one stream, I use two NetStreams and end up with two .f4v files - one with the video and one with the audio. I need to now combine these into one proper video file. However, these files don't import into Premiere (which is really lame) - giving an unsupported compression error or some such nonsense. Can someone please let me know how I can combine these files, and why the heck Adobe's own products can't open the files. Also, I use Windows 8 and the Camera app (metro app) can record 1920x1080 video with great quality audio, at 30fps. FMS cannot come even close to this... WHY? Really wish Adobe would fix this.

I am attempting to login to our AMS Admin Console but I've clearly lost the only login credentials we used in the past.We had a developer set up AMS for us but he has since gone completely MIA. Is there a way to edit a file on the server to be able to login again?I am no developer but I can't even seem to find the AMS application files on our server. I'm really quite confused as you can see.Any guidance?

HelloOk, so i've setup AMS and FME for live HDS multi-bitrate streaming.  Everything is working fine, except for the player will only play the lower bitrate stream.  When the player first starts it starts at the lower bitrate, then goes up to the highest, but only stays on the highest for a few secs, and goes back and stays on the lowest bitrate. I know the first question is about bandwidth, and there is plenty of bandwith on the player pc and server. I can also see that all 3 streams are being created in the streams folder.  There must be something im missing.  Any help is greatly appreciated.  Event.xml:<Event>  <EventID>liveevent</EventID>  <Recording>    <FragmentDuration>4000</FragmentDuration>    <SegmentDuration>16000</SegmentDuration>    <DiskManagementDuration>.05</DiskManagementDuration>  </Recording></Event>Liveevent.f4m

is it only can use in swf? not can use air like in android,ios?

I am just wondering about the experiences with Adobe Media Server on Linux or Windows.What is easier to configure for a conference calls application?Your help will be greatly appreciated.Thanks,LiveComLinks NL,/Hans

Content provider set his encoder to stream content on my Intake FMS 4.5 server. Is it possible to have an incoming rtmp stream to FMS server be kind of duplicated/feeded/relayed to other FMS servers? So that I'll have, let's say, 5 intake servers that are streaming to their own edge servers.

Using Wirecast 5.0.1 to stream to a Flash Media Server. Have experienced occassional video freeze and drops. Wirecast log contains several instances of the following string:"INetStream queueLength over max = 1489". Is this a message sent back to the client from the server and exactly what does it mean and what can or has to be done to correct it?

Last week I stumbled across the checkBandwidth() API between FMS/AMS and clients: http://help.adobe.com/en_US/FlashMediaServer/3.5_Deving/WS5b3ccc516d4fbf351e63e3d11a0773d56e-7ffa.htmlWhen I tried this out with apps that basically just do nothing but poll that, I get erratic readings all the time.  This is between different clients, different networks, and different versions of FMS/AMS.  The documentation for AMS 5.0 isn't even much different from the documentation for FMS 3.5, which was in that link above.  The readings tend to come back saying anything at all between 0 and maybe 140,000 Kbps, constantly changing from one extreme to the other, and to everything in between.  Sometimes they'll just sit at 0, when I know they have very good, very stable upload and download speeds and are able to communicate with the FMS quite well.  This is wrong behavior.I would just write this off as something that just doesn't work, but like I said, the documentation between F

Hi. for more than 3 days i'm trying to build multi server's cast Based on the attached model.Everything I've tried have not helped me.can you kindly help me to build the right " vhost " file for each server as described in the attached model ???I really appreciate itThank you very muchElad.----- THE VHOST -----<VirtualHost>          <!-- This tag contains a list of <Alias> tags. -->          <AliasList>                    <!-- You can specify additional names by which to connect to this -->                    <!-- virtual host by specifying the name(s) in one or more Alias  -->               &nbsp

Hi,I have my own FMS server its mainly for HDS streaming. Also I have edgcecast CDN. I wasing using WoWZa earlier and restreaming through edgecast FMS. I need experts advice for my HDS streaming setup with my FMS server and edgecast CDN. I used Adobe Flash Encoder and then HDS stream out put from my server is passing through Edgcast as my CDN. Any other most recommended methode for best perfomance as with following setup:-Flash Encoder >> FMS server >> Edgecast CDN >> Players I means is it Edgecast just as caching or restreaming FMS source ?Also if its restreaming FMS should have produce RTMP source, right ?thanks in advance,Regards,Navas

OK, so I have built an OSMF player and the brief was to implement closed captioning. So far, so good, I went to here:http://help.adobe.com/en_US/adobemediaserver/devguide/WS5262178513756206-55daa065139e25f4596-8000.htmlAll worked well enough using a SerialElement to schedule content. but then the brief was to implement a new style of advertising and...Midrolls may or may not be inserted at unknown points and, with a single file for my main content, the only logical way to implement it is to pause within a ParallelElement. That way, the advertising can overlay ads as MediaElements at appropriate points and stop / start / play / pause to its merry satisfaction.But...OSMFCCDecoder class breaks with ParallelElement.It simply does nothing.There is no way to specify a clip to listen to. No way to get it to show anything.Therefore, I can either have closed captioning or advertising and, unsurprisingly, my customer wants both...Is there any way to get sources for OSMFCCDecoder or, better, a mo

Hi,We need to purchcase adobe media server extended verion but the plan and price are not available online. We have tried online contact forms and given number but no one responde to them.Does anyone have some contact from where we can have information.

Hi,I've been trying MPP for the last few days and it works fine for single stream but when I have 3 stream, at different bitrates, pnly the last one works, other work for short period and then they stopHere is the code I am using. Please guide as to how I can change this code to republish more than one stream!// Called when the client publishesapplication.onPublish = function(client, myStream) {    trace(myStream.name + " is publishing into application " + application.name);    // This is an example of using the multi-point publish feature to republish    // streams to another application instance on the local server.    if (application.name == "live/_definst_"){        trace("Republishing the stream " + myStream.name + "into /livepkgr2");        nc = new NetConnection();        nc.connect( "rtmp://localhost/livepkgr2" );&n

HI, we have signed up to use Amazon Cloudfront via their Sydney server.  However, your fees do not include Sydney...http://www.adobe.com/au/products/amazon-web-services/pricing.htmlCan you advise?Please also explain what "100 RTMFP" means (in particular, the trailing 'p'? 😉thanks,Geoff

Hi,On monitoring the x-duration field of the access logs I find the value 0 for both client-pause and client-unpause events. These two are stream level events. According to the definition for x-duration atleast any one of these two fileds should have non-zero value.x-durationDuration of a stream or session event. For a stream, this value is the number of seconds the stream has played. For a session, this value is the number of seconds the client has been connected.Is my interpretation wrong ? Or there is something wrong with the logging mechanism of our AMS. Kindly give your comments.Thanks in advance.Veera

When using RTMFP multicast is working , iam able to see in router , but when viewing stream at remote end on satellite TCP packet are generating instead of UDP .Scenario : Using Encoder with RTMP ---then Media server with Fusion RTMFP--- then Router for multcasting ---Remote end Router---HUB Satellite ---Remote Satellite ---remote End computer on which using multicast player--- 

Hello guys, when will AMG support G711a codec? Apart from that, is there an option of getting this codec's DLL somewhere free or commercially?

Hi,I live a stream to the application "livepkgr" with H.264 and G.711(come from IP camera), I can get the video and audio from player, but there is only video data in the .f4f file, no sound data.and in the .flv file(when with "record"), there is only audio data. Doea anyone know why this happened?Should I set some configuration or something?Thanks.

Hi,We have a problem to do HDS streaming in my FMS ...attached the screenshot showing error when i tried in my sample video player.Normally what could be the reason for it. RTMP is working without any probelms.Please advice where I can check the errorlogs for this issue.

Hello,I want to log messages when user starts recording on, so i can check  and debug errors in recording.I use this code - application.onAppStart = function() {      application.allowDebug = true; }in main.asc and then restart media server. there are not any differences between previous logs and current logs after recording.It seems AllowDebug is not working.can somebody help me please?Thanks