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ririck70
Participant
May 6, 2008
Answered

FMS 3 Questions for live video conferencing

  • May 6, 2008
  • 1 reply
  • 364 views
We have some basic questions for an installation (windows 2003/FMS3) server for a videoconference applicationt with max 10 rooms of 3 persons x room connecting remotely:
1) Is there an upgrade for the fms components optimized for FMS3? If not should we go with the old ones?
2) is there a way to detect client bandwidth and latency and optimize Audio/Video based on that on a per client basis?
3) We are experiencing hight latency issues that are not acceptable for videoconference (5-10 sec): is there a way to optimize server config for that? Is there a way to have Quality Of Service configured on the network like for Voip apps?
4) I'we read in the manual that 44hz encoding is faster than lower rates for audio? I would have expected the contrary. As the audio is the most important for our system is there a way to optimize that?
We have big echos (also with the reduce echo settings) and bad quality.Any way to improve this?

    This topic has been closed for replies.
    Correct answer KevinStreeter
    I'll try to answer a couple of these...

    1) The current FMS components will work FMS3, although certain features (like redirect) are only supported in the lastest playback component. If everything is working great for your app, then don't feel like you have to upgrade.

    2) FMS3 has the same script-based bandwidth detection capabilities as FMS2, as well as a new, more efficient "native" bandwidth detection feature. Check out the FMS3 live docs for how to use this feature. Latecny detection is not supported out of the box, but you could easily write SSAS to gauge round-trip times within a few seconds accuracy.

    3) Unfortunately, there is nothing that can be done on the server to overcome a high-latency link. Real-time communications will always "lag" over this kind of connection. (Have you ever watched a live satellite feed on TV?)

    4) I don't have any specific details about the encoding speed for different sample rates, so I won't comment on this. I will say that, in general, encoding does not introduce significant latency (unless your client happens to have a *very* slow processor). Its network latency and network congestion that will do the most harm. For the best experience, try to keep the total bitrate of the streams as low as possible, and be sure to use bandwidth detection,etc to ensure you are never exceeding the capabilities of the client's connection.

    1 reply

    K
    KevinStreeterCorrect answer
    Participating Frequently
    May 6, 2008
    I'll try to answer a couple of these...

    1) The current FMS components will work FMS3, although certain features (like redirect) are only supported in the lastest playback component. If everything is working great for your app, then don't feel like you have to upgrade.

    2) FMS3 has the same script-based bandwidth detection capabilities as FMS2, as well as a new, more efficient "native" bandwidth detection feature. Check out the FMS3 live docs for how to use this feature. Latecny detection is not supported out of the box, but you could easily write SSAS to gauge round-trip times within a few seconds accuracy.

    3) Unfortunately, there is nothing that can be done on the server to overcome a high-latency link. Real-time communications will always "lag" over this kind of connection. (Have you ever watched a live satellite feed on TV?)

    4) I don't have any specific details about the encoding speed for different sample rates, so I won't comment on this. I will say that, in general, encoding does not introduce significant latency (unless your client happens to have a *very* slow processor). Its network latency and network congestion that will do the most harm. For the best experience, try to keep the total bitrate of the streams as low as possible, and be sure to use bandwidth detection,etc to ensure you are never exceeding the capabilities of the client's connection.

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