Ben Timms 18.Jul.12 09:06 PM a Web browser Audio/visual services8.5.2 IFR1Windows
We are about to deploy desktop video conferencing using SUT Lite with a Polycom bridge and we'd like to apply Quality of Service.
We'd like to QoS both the connection to the SIP Proxy/Register on the Media Manager (TCP 5080), as well as the data stream. I believe is RTP embedded in UDP packets from 20830 for audio, 20832 for video). I think the range is 100 ports from this, and it gets incremented by 2, but I can't find where I read that again.
I've done a netstat -an with and without a call, and from what I can see the difference is:
These ports seem to be the same regardless of how many participants or location (limited testing).
This was an audio and video call, does this mean 2 ports are used for audio (incoming and outgoing), and 2 ports are used for video (incoming and outgoing) ?
We want end-end QoS so we'd need it on the connection from the Media Manager to the Polycom DMA then onto the RMX and back.
I think it would be like this for multipoint calls:
Client -> Media Manager -> Polycom DMA -> Polycom RMX -> Polycom DMA -> Media Manager -> Client
and for point to point calls:
Client -> Media Manager -> Client
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How to apply QoS for Sametime / Pol... (Ben Timms 18.Jul.12)
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