It is indeed not for the faint of heart.. that being SonicWall traffic shaping for RTP.
QoS in general won't help unless your LAN is congested.. which my largest clients may have some link aggregation between switches to relieve such congestion anyway.
Most ISPs won't do QoS with you anyway. We have gone with some ISPs who provide SIP trunking services and have had relative success.. at least with browbeating them for any WAN traffic contention since they are delivering it.
Our standards include a dedicated IP, a non-SonicWall NAT router (we go with a NetGear model..), and "not" Comcast or a provider with high jitter after measuring for jitter.
While we can succeed with a SonicWall on Comcast with a single IP just fine.. too many problems with other sites.
It's like the whole T38 fax option. Sure, I got it to work between an MFP from one client to send/receive fax to the 3CX Fax service or a fax connected via ATA.. but otherwise was inconsistent or the handshakes would fail (I could hear the attempt at the far end.. oh, the sysop days of BBS..).
We developed those standards based on painful experience.. BEFORE that training conference up at Redmond in May. Wasn't a surprise to hear SonicWall was.. not recommended.. yeah, let's put it that way.
Why the dedi? We still have SonicWalls with VPN licenses for our non-SBS or Mac clients.. so we have two routers on site (not the ideal topology, I know..).
I found two years ago after an associate left me holding the bag with a 3CX site which audio was one way.. inconsistently. Though the access rules and NAT policies called out the RTP ports for both inbound and outbound NAT.. the SonicWall decided to utilize those ports on the WAN regardless.. which killed the inbound audio. After I set those rules/policies to translate on a different WAN IP, their problems disappeared. That started the "dedicated IP" requirement.
Another client had the "dedicated IP" but audio quality would suffer. With some of the UTM features enabled, two site-to-site VPNs, with client IKE and SSL VPNs, and finally with the WLAN option built in (TZ200W..), we monitored and saw the processor utilization peg at times and concluded the appliance could not keep up at all times.. which is when the call audio suffered.
So, yes.. it can be done. It is a bit of a process to step through, however. Indeed, I would suggest a dedicated NAT router and a dedicated IP on your circuit. I would run some jitter tests against the circuit, though, to make sure the circuit is suitable for VoIP at all to begin with.. first. For some, we've gone with the cheap DSL option. Makes for a PSTN line for fax, too!
