Well, if your SIP provider will provide you with SIP URI, I guess you could bypass the Asterisk server. But, I am not sure that they do.
Another thing to note is the ping time to voipcheap’s sip server. Of course, my pings are from Montreal… it really depends where you are pinging from. But never the less…
Pinging sip.voipcheap.com [126.96.36.199] with 32 bytes of data:
Reply from 188.8.131.52: bytes=32 time=164ms TTL=231
Now lets ping the servers I use, just as a point of comparison:
Pinging sw1.link2voip.com [184.108.40.206] with 32 bytes of data:
Reply from 220.127.116.11: bytes=32 time=92ms TTL=42
Pinging sw2.link2voip.com [18.104.22.168] with 32 bytes of data:
Reply from 22.214.171.124: bytes=32 time=94ms TTL=42
Pinging west.voipjet.com [126.96.36.199] with 32 bytes of data:
Reply from 188.8.131.52: bytes=32 time=69ms TTL=43
Pinging east.voipjet.com [184.108.40.206] with 32 bytes of data:
Reply from 220.127.116.11: bytes=32 time=68ms TTL=43
Does that 70ms make a difference? Well, ever little bit is just another strain on the system so… YES.
If you have some sort of ping app, you might want to try sending out some pings from your phone to your sip servers and see what kind of ping times your getting to start!
The qualify thing is used so that you can run behind NAT/Firewall, correct me if I am wrong, but I believe it just keeps the connection from the SIP client to the SIP server alive, so that if there is a call to a DID and it comes in through the asterisk server, then it has an idea of where to send it rather then it just getting obliterated by a NAT/Firewall.