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I have the following situation: 2 instances of FMLE on one machine and another machine with AMS Starter.The instances of the FMLE are:1. Video H.264 + audio MP32. Audio only MP3 - track 1The application on AMS is a livepkgr/_definst_ and i am using HDS with a live eventI have an OSMF based player to consume the streams and to do audio switch using late binding audio. When I consume the Video with the Default audio everything works just fine. The problem is that when I try to consume the Video with the other audio track, the audio tack has a 1 second delay compared to the video.I have tried to fix this by changing the configurations on the FMLE, but it doesnt seem to work. So I really want to know how does exactly AMS syncronize differents streams, in this case a video stream with a different audio stream.
A new Adobe Media Server release v5.0.3 is now available for download on the updaters page. If you are using PHDS/PHLS or PRTMP this release is a must download for you. Content Protection certificates included in the previous releases are expiring on 10th August so this new release refreshes the following certificates:• creds/phds_license_server.der• creds/phds_production_transport.der• creds/phds_license_server.pfx• creds/phds_production_packager.pfx• creds/phds_dataFor the list of other enhacements and bug fixes included in the release please refer to release notes available herehttp://www.adobe.com/support/documentation/en/adobe-media-server/releasenotes.html#ams503Adobe Media Server 5.0.3 direct download link is herehttp://www.adobe.com/support/flashmediaserver/downloads_updaters.html
Hi All,I have developed a video chat application and during my testing, I found that if video chat is going on and in between the call if some one unplugged Internet cable then FMS admin console is taking too much time to disconnect as well as remove from list.Is there any way to remove disconnected client immediately from admin console then please help me either by doing any configuration or logic.Thanks Ram
Hello, first sorry for my bad English, I speak French ...I succeeded install fms 4.5 on my centos 6.3 server. By against I had apache so I did not install apache fms included. On my centos server, I already whm and cpanel. I do not understand how to configure my existing apache to run with fms. And I do not know at all how to check if my fms installation is correct, because I can not access the administration console .... Will you explain to me a little please?
Hi,I have installed AMS on a system and trying to connect to "live" streaming from another pc. It is working from same but not from another.Following is the rmtpnc.connect("rtmp://122.160.124.114/live");but vod is accessable. Do i need to check any setting.Thanks,
i need to know how many conections can i have with a license of adobe medie server proffesionali need some help. thanks
I am trying to set my streams locked to only certain domains. As long as I leave it wide open my player plays the content fine. Once I set a domain I get "We are having problems with playback. We apologize for the inconvenience. Below is my current cross domain file.<?xml version="1.0"?><!-- http://mydomain.com/crossdomain.xml --><!DOCTYPE cross-domain-policy SYSTEM "http://www.macromedia.com/xml/dtds/cross-domain-policy.dtd"><cross-domain-policy> <allow-access-from domain="*www.mydomain.com" /></cross-domain-policy>
I am looking to create an application that would take an incoming sip call (H264 and G.711) and convert that to a pHLS stream. Does pHLS support the g711 audio? If not, Is there a way to transcode it and is there any examples?Kyle
Quick and short....What is the best way to stream to balckberry and Windows Phones?Thanks!
Is Apache necessary to stream content?
hi everybody first of all i'd like to apologize for my english . it's not perfect because i'm still learning .thanks in advance.well i have a question and until today i didn't find an exact answer even with google and in forums.in the adobe media server comparison ( http://www.adobe.com/products/adobe-media-server-family/buying-guide-comparison.html ) it is said that adobe media server 5 extended has a limit of 15000 RTMFP P2P INTRODUCTIONS.i'm not able to figure out what does that mean ????????? * does it mean that only 15000 users can connect to adobe media server via rtmfp and after they can connect each other ? or * does it mean that only 7500 couple of 2 users can be connected ?i'd like to know exactly what this limit stands for ??????????me and some friends are making a real time chat with webcam and audio and we plan to buy adobe media server 5 extended even tough the price is expensive !!!!!!!!!! that's why we want t
When try to change the .asc extension to .ase far.exe has prompt error. same time when run far.exe for server 3.5 on 32 bit machine it works fine.My observation when i am using For each it fails too for server 3.5 for 32 bit machine.thanks in advance.Vishal Sood
Hi I'm new to FMS/AMS I am creating a video recording app and I'd ideally like to pull down the resulting video file and post it to an S3 bucket.I know WOWZA can do it but if possible I'd like to stick with AMS. Is it just me or is it a bit of wizardry to actually get useful info in the docs on beginning with ams re writing applications etc?!I was wondering if it's possible to pull a recorded file down and do something to get it as a byte[] or the like and POST the file elsewhere? Rather than streaming it down over rtmpThanks,Tom
I'm trying to capture audio from a mic and save on an AMS server running AMS Standard.I can make a connection without problems with "rtmp://server_name/live", but when I try to start recording audio with mic = Microphone.getMicrophone(); mic.rate = 44; mic.gain = 100; mic.setSilenceLevel(0); try { nsPublish = new NetStream(nc); nsPublish.addEventListener(NetStatusEvent.NET_STATUS, netStatus); nsPublish.attachAudio(mic); nsPublish.publish("recordTest", "record"); } catch(error:Error) { setStatus(error.toString()); return; }I get the following error: Application doesn't have permissions for server-side record/append of streams; access denied to stream recordTest.Can anyone tell me why and how to fix
I have a video player client accesing constantly-updating videos on the server.When one of these files are deleted, or changed, the player doesn't notice, even through a browser refresh, and keeps playing the old file for a few mintues.How can I make it so that this doesn't happen?
Hello, we are new to the Flash Media Server world. We are using this with Amazon EC2 integration. We currently have a need to stream multiple streams from various locations at the same live event. We have 3 stages we need to stream at a music festival. How can we utlize streaming these different stages to one EC2 instance running Flash Media Server? Currently it seems we are only able to generate one .m3u8 file with the cloudformation template. Ultimately we'd like to have a playlist in our end player where our subscribers can access the multiple stages. It seems this should be possible. Any tutorials on how to do this with Amazon EC2, Cloudfront and Flash Media Server?
Hi,I am trying to publish live stream from Adobe Flash client that can be viewed on iOS device. One known issue is that the Flash client does not support AAC audio encoder when it publishes a stream, but AAC is required for iOS viewing. Thus I am using ffmpeg to transcode the audio to work around this issue. The setup looks like following: Publish(h.264+Speex) Pull Re-Publish(h.264+AAC) Flash Client Publisher -------------------> AMS -----> ffmpeg ---------------------> AMSThe command line I use for ffmpeg is like this:ffmpeg -re -i rtmp://xxx.xxx.xxx.xxx/livepkgr/orgstream -c:v copy -c:a aac -strict -2 -f flv rtmp://xxx.xxx.xxx.xxx/livepkgr/livestreamThis se
same as above.
Hello, I want to play my video files (which are on s3 bucket ) like live .i.e.mox.tv i want play videos on my website.Some buddy please help me ..for that what i need to do?my video are in mp4 formate with h.264 codecsMessage was edited by: lakhanp22
In the default setup of FMS 4.5, the video files for Flash streaming and the video files for HLS streaming sit in two separate directories. Is there any reason why I should not adjust my config settings so both functions point to the same directory? Essentially, I would like to make the "applications" directory and the "webroot" directory the same directory. I would then only need to publish one MP4 file on my server. Flash streaming and iOS streaming would both be using the same file.Thanks.
I am selling this domain. As FPS can stand for frames per second, I thought this would be an ideal forum to provide the users here a chance to purchase a rare 3 letter domain.This is totally legit as you can see from my provide, mods please don't close or delete, thanks.
Can you call a function on an Adobe Media Server application with anything other than Flash?I have a `.asc` file up on a server which controls the creation and deletion of files.Right now, I can only interact with my server by using the AS3 method `NetConnection.call()` from a client Flash file.Is there a way to access these methods through, say Javascript or anything else?
I created a stream connecting to FMS4.5 and is able to Smart Seek with setting inBufferSeek=true, detail's as below:ns=new NetStream(nc);ns.inBufferSeek=true;ns.backBufferTime=60;ns.bufferTime=3;ns.play("mp4:abc.f4v");it's ok to pause at 5's while playing fine. when i want to backward frame i use ns.step(-1); //workit works.however, when i firstly pause the stream and then seek to any time point, eg:ns.pause();......ns.seek(20);After recieving NetStream.Seek.Complete, when i tried to backward/forward frames, it does not work at all. eg:ns.step(-1); //not workns.step(1); //not workIf i run ns.resume(); ns.pause(); it was able to forward frames but not backwrad. eg:ns.pause();......ns.seek(20);......if(e.info.code=="NetStream.Seek.Complete"){ ns.resume(); ns.pause();}......ns.step(-1); //not workns.step(1); //workSo my question is, how to make it work to backward frames after that seeking has completed ?or, is there any possibility that this issue coul
Need Help pleaseHas anyone ever connfigured a Live stream coming form RTP or RTSP source to RTMPWould like to convert or inject the RTP source into AMS so i can rebroadcast.
I'm new to AMS. I have the Starter version at the moment. I have three basic questions:(1) The only thing I need at the moment is audio capture from a mic. What technology is invoved in this? RTMP, HTTP streaming, RTMFP... there are too many of terms.(2) If I want to allow for a maximum of 100 simultaneous audio capture sessions, what version of AMS would I need?(3) I'm trying to configure an external install of Apache (on linux) to work with AMS, but is struggling without good documentation. Does Adobe have something for me to look at. A sample httpd.conf for CentOS would be wonderful.Thank you.
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