Need an intro to QoS on Cisco

I am about to add VoIP service behind a 871W router.

So I will likely want to add QoS on the router to give priority to the VoIP packets.

Is there a quick intro on how this is done on Cisco gear ? (general concepts, commands used, are there queues, do you define maximum throughput and a minimum trhoughput given to a class ? Are classes defined based on a port number use for TCP/UDP streams ? Or based on IP address of a host, or looking into packet contents to detect SIP traffic for instace ?)

Can this be applied to both inbound and outbound traffic ? The router can make obvious choices for outbound traffic.

Can the same be done for inbound traffic, forcing router to drop some "internet" packets to cause those flows to slow down, leaving sufficient bandwidth for the VoIP traffic ? (I assume this would require the router be told of the available bandwidth for inbound traffic so it knowsn when lower class traffic needs to lose packets ?)

I have done a bit of this and I found the documentation sorely lacking (say 4 years ago).

Trouble seems to be that there are many platform dependencies and they are not very clearly documented.

I got the impression that it was basically all smoke and mirrors and effectively did not work as advertised except in some particular cases.

Inbound, my plan was:- Do rate limiting on the non voice OUTBOUND traffic on the INSIDE interface. Leave enough headroom for the voice. Thsi will work well if the non-voice traffic is TCP but if UDP then it may well be of no use at all.

TCP covers http traffic but not necessarily audio/video.

Outbound - I can offer no real advice since my experience was all on 877 and the ATM int seems to have weird and wonderful QoS configurations.

If you have support open a case and let someone else worry about it.