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QoS + VOIP, SIP client phones to Internet server

So I've read and read, and read some more all sorts of different posts and none of them are real clear.  QoS on the UTM seems to be all about throttling and limiting.  Is there a way just to use QoS prioritization like cheap routers/firewalls will and all the other commercial ones will?

I have 4 Cisco SPA525G VOIP phones using SIP + RTP to communicate to my cloud hosted "pbx".  If we're not using the internet everything works well.  However, if I decide to upload a file to dropbox or exchange or whatever, the phones degrade horribly, downloading seems to not affect it as much.  I have an 18Mbit down, 1.5Mbit up connection.  According to my VOIP provider, "marking OSI Layer 2 packets with high-priority (5) class tags (802.1p and IP Precedence)", and they tag all of their voice packets with DSCP value of 46. 

So what settings are required to give these packets absolutely priority over everything else.  I shouldn't have to set guaranteed bandwidth with priority being given to them.  If priority is utilized it should just put them to the front of the line no matter if I have 1 phone in use or all 4 of them.

EDIT: SG135W running 9.412-2 if that matters.

Thanks!



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