I'm using windows media services 9 to provide streaming service and would like to make sure users can only play within specified interval to prevent abuse. The media server reads and from asx file and the player can start and end at the correct offset. I can get the start offset at AuthorizePlayEvent using
IWMSContext.GetQwordValue(WMSDefines.WMS_PRESENT_START_TIME,WMSDefines.WMS_PRESENT_START_TIME_ID, out dec, 0);
But I'm having problem getting the duration or end offset. I know windows media server writes log that contains c-starttime and x-duration. But the x-duration here doesn't seem to be what I want. I like to detect the that gets passed from asx file at either AuthorizeOpenEvent or AuthorizePlayEvent. Is this possible?
Thanks!
I ended up rolling my own synchronized dictionary to keep track of active players at every heartbeat and disconnect player that exceed specified interval. I spent around 2 days looking for build-in server side parameters and was surprised to find nothing is available reliably.
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In my Codename One app, I need to limit the duration of an audio capture to few seconds. I call Capture.captureAUdio() for that and there does not seem to be an equivalent to Capture.capturePhoto(w, h) that could bound the captured audio to a given duration.
What I've done so far is measure the audio capture duration and if it is too long then ask the user to start again if they will. But I find it a bit cumbersome.
Is there anything better that can be done ?
Any hint appreciated,
No.
For audio though you can use use the media recording feature from MediaManager which isn't as intuitive as Capture but allows more "fine grained" control.
I am new to sip protocol,i went through the basics and have these following doubts
1)In registering process when i captured using wireshark,i figured out that from and to headers are same when i read rfc 3261,it says that "to" header indicates whose registration is to be done and from" indicates person responsible for registration.The to and from fields are same unless it is a third party registration.it is not clear to me,how can it both be same and what is a third party registration.
2)Does sip have any keep alive mechanism,in zoiper we have the option of giving expiry time (3600 default),but for registration it is 70,for subscribe it is 60 and for invite it is 3600. how these values are automatically selected?
3)The user agent finds registrars using configuration.dns look up and multi-casting.In what case multi casting is preferred,pls explain the method also
what i did was ,installed an asterisk server ,zoiper applicationregister msg capture is attached,created a zoiper account,captured using wireshark in loop back mode.attaching screenshots of captures.Thanks in Advance
Regarding to and from fields in REGISTER:
The "from" field here is just a logical field which should not be checked. If differs from the "to" field that means that "from" registers in name of "to".
But I can't think of any scenario when this should be checked (maybe it can be used for something -app specific- in some complicated scenario). You should just follow the usual authentication process (digest auth or other) and skip this field.
Regarding point 2 (expiry time):
Your mentioned settings in Zoiper are just arbitrary.
Low values (below 200) can be used if client or server doesn't support NAT keep alive (via NOTIFY or simple \r\n\r\n messages). In this case the REGISTER message will keep alive the UDP binding in NAT routers.
Higher values can save some server side processing work and CPU resources
I usually recommend a 600 sec expire timer and 40 sec NAT keep-alive messages.
For INVITE the expire field actually means maximum ring time and it is rarely used.
Regarding point 3 (finding registrars):
The SIP server (registrar server) is usually entered manually in client configuration or set by auto-provisioning. If the server is on the same LAN, then you might be also to detect it also by multicast but this is rarely used.
Here is a good tutorial.
I'm trying to create sort of listener. I need voice xml that will keep the call open and not hang up. I tried to do that with tag, but max i can get is 2 minutes. Is there some sleep or loop tags with time out I can use?
Have you looked at VoiceXML's close cousin Call Control XML (CCXML). CCXML provide more call control (as the name implies) and allows you to kick off VoiceXML sessions from it. It can monitor the call for the duration. What are you listening for? If you provide more information on what your solution is supposed to do I may be able to provide more specific guidance.
In Amazon Web Services, their queues allow you to post messages with a visibility delay up to 15 minutes. What if I don't want messages visible for 6 months?
I'm trying to come up with an elegant solution to the poll/push problem. I can write code to poll the SQS (or a database) every few seconds, check for messages that are ready to be visible, then move them to a "visible queue", or something like that. I wish there was a simpler, more reliable method to have messages become visible in queues far into the future without me having to worry about my polling application working perfectly all the time.
I'm not married to AWS, SQS or any of that, but I'd prefer to find a cloud-friendly solution that is stable, reliable and will trigger an event far into the future without me having to worry about checking on its status every day.
Any thoughts or alternate trees for me to explore barking up are welcome.
Thanks!
It sounds like you might be misunderstanding the visibility delay. Its purpose is to make sure that the polling application doesn't pull the same item off the queue more than once.
In other words, when the item is pulled off the queue it becomes invisible for a predetermined period of time (default is 30 seconds, max is 15 minutes) in case the polling system has a cluster of machines reading from the queue all at once.
Here's the relevant documentation:
http://docs.amazonwebservices.com/AWSSimpleQueueService/latest/SQSDeveloperGuide/IntroductionArticle.html#AboutVT
...and the sentence in particular that relates to my comment is:
"Immediately after the component receives the message, the message is still in the queue. However, you don't want other components in the system receiving and processing the message again. Therefore, Amazon SQS blocks them with a visibility timeout, which is a period of time during which Amazon SQS prevents other consuming components from receiving and processing that message."
You should be able to use SQS for your purpose since you can leave an item in the queue for as long as you want.
7 years later, and Amazon still doesn't support the feature you need!
The two ways you can sort of get it to work are:
have messages contain a delivery target datetime in their message_attributes, and have the workers that consume the queue's messages just delete and recreate any message that is consumed before its target, with delay = max(0, min(secs_until_target_datetime, 900)) ; that would allow you to effectively schedule a message for any arbitrary future time;
or,
(slightly less frequent and costly:) similarly, if a message isn't due to be handled yet, recreate it and change its visibility timeout to be timeout = max(0, min(secs_until_target_datetime, 43200))
The disadvantage of using visibility timeout is that any read will re-trigger it.
There has been a direct AWS solution possible since 2016-12-01: AWS Step Functions
Each execution can last/idle up to one year, persists the state between transitions, and doesn't cost you any money while it waits.
I'm attempting to use a large number of short sound samples in a game I'm creating in Silverlight 2. The samples are less than 2 seconds long.
I would prefer to load all the audio samples onto the canvas during the initualization. I have been adding the media element to the canvas and a generic list to manage it. So far, it appears to work.
When I play the sample the first time, it plays perfectly. If it has finished playing and I want to re-use the same element, it cuts off the first part of the sound. To play the sample again, I stop and play the media element.
Is there another method I should use the samples so that the audio is not clipped and good performance is obtained?
Also, it's probably a good idea to make sure that all of your audio samples are brought down to the client side initially. Depending on how you set it up, it's possible that the MediaElements are using their progressive download functionality to get the media files from the server. While there's nothing wrong with this per se (browser caching should be helping you out after the initial download), it does mean that you have to deal with the browser cache, and there are some potential issues there.
Possible steps to try:
Mark your audio files as "Content". This will get them balled up in the .xap.
Load your audio files into MemoryStreams (see Application.GetResourceStream method) and call MediaElement.SetSource().
HTH,
Erik
Some comments:
From MSDN:
Try to limit the number of MediaElement objects you have in your application at once. If you have over one hundred MediaElement objects in your application tree, regardless of whether they are playing concurrently or not, MediaFailed events may be raised. The way to work around this is to add MediaElement objects to the tree as they are needed and remove them when they are not.
You could try to seek to the start of the sample to reset the point currently being played before re-using it with:
mediaelement.Position = new TimeSpan();
See also MSDNs MediaElement.Position.
One techique you can use, although I'm not sure how well it will work in Silverlight, is create one large file with all of your samples joined together (probably with a half-second or so of silence between each). Figure out the timecode for each sample and seek the media element to that position and play. You'll only need as many media elements as simultaneous sounds you want to play.