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3) Most routers mark packets outbound, and little emphasis is placed on inbound marking. This is because by the time the packet gets to you, unless YOUR router is saturated the packet will get through with low latency.
"outbound" depends on your perspective.
"YOUR router is saturated" (If your router is saturated, I would recommend drying if out.) Usually the links that routers are connected to become saturated.
Marking the pakets on the way in may or may not happen, but the net result would be the the same. (Unless your WAN (outbound) connection was faster than your LAN (inbound)).
Further,
VoIP is UDP based, and is highly sensitive to latency. The Internet is a place where latency is highly unpredictable and the more network hops (the further geographically) your packets have to travel, the higher the end to end latency will be; as such, VoIP is likely to remain a low quality voice transport for a while. Contrastly, your analogue telephone line, when you make a call from US to China, actually reserves an entire set of *dedicated* DS1 (64Kbits/sec) analogue pipes from one end to the other. In other words, there is zero sharing; hence the guarantee and high quality.
Actually you get less than 64Bbit/second dedicated if your telco is in the US.
Google "Robbed Bit Signaling"
VoIP is UDP based, and is highly sensitive to latency.
Bad generalization there. RTP is UDP, but not all VoIP protocals use RTP. I assume you understand that while SIP is a VoIP standard, the standard for VoIP isn't SIP.
Perhaps one day, when all the major Telcos and ISPs have more pipe than they know what to do with, long distance VoIP will come close in quality to analogue phones... until then it's a complete crap shoot. You might get amazing quality to some locations on some days, at certain times 99/100 times, and to other locations 80/100 times the VoIP call is utterly useless.
The setup and codecs I use actually exceed carrier quality "G711" codecs. If you aren't an expert, don't try to sell yourself as one.
In resume, you can tweak your home router all you want. It might help slightly since your router would become a saturated network point due to you using bitorrent simultaneously; however, the other 8+ hops to get to "China" are completely out of your control.
Like I said before. The "router" isn't getting saturated. Why are you pushing this fallacy?
Who is calling China via VoIP? Why would you even mention china? My IP phones register with an asterisk server in texas. I can handle the 20 milliseconds, and so can sensitive UDP packets.
My recommendation is that if you have a say 1Mbit Up/Down pipe for broadband internet; that before you make your VoIP call, that you throttle your bittorent software (in the software itself) to use only 850Kbits up/down. VoIP protocols can suck up anywhere between 8Kbit/sec (highly compressed) to 110 Kbits/sec (uncompressed). So by leaving 150Kbits for VoIP, there's a good chance the VoIP and torrents can co-exist peacefully.
Cheers,
ADeptus