some of our customers are experiencing Audio Quality Issues with CWMS PC-VoIP.
It is clear, that this is dependant from where users join the Meeting, nevertheless we have observed with our customers, that using the same connection and same Audio Equipment, a Jabber Softphone call will deliver a better Quality experience in most cases. (I personally feel the same, and am therefore myself using Jabber-Calls instead in most of my meetings).
Therefore I have two Questions:
1) Can PC-VoIP Client be prioritized via "normal" marking based on Port Usage?
As per the Guides CWMS can mark the Audio/Video Traffic with proper DSCP Values, nevertheless as per my understanding, this only works "one way" from CWMS to Client, not vice versa. Therefore it would be interesting if it for example makes sense to prioritize UDP Ports 9000, 9001, 9002, 9003, 9004, 9005, 9006, 9007, 9008, 9009 from/to CWMS.
Which information is exchanged via this Ports? Is it only Audio/Video, or does it as well include Screensharing, etc.??
Any other possibilities to prioritze audio in PC-VoIP?
2) I have read, that cloud based WebEx uses OPUS Codec as default since Q4-2014. Is this also the case for CWMS?
If yes, since which Release?
If no, is it planned to include OPUS, so we can communicate some "progress" for PC-VoIP Quality?
I am not sure where you are having most issues, It appears you can not do it at CWMS level, but you can certainly catch this traffic and priortise at network level. There are couple of ways you can use audio port matching or NBAR type protocol based matching at your edge routers, marking and priortising RTP audio traffic.
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