I would like to use tc to control jitter for an RTP streaming application. I am OK with introducing latency if it is a means to the ends of minimizing the jitter experienced at the streaming application.

I believe I understand how tc-tbf / tc-htb is used to shape traffic, but I am having trouble visualizing how I use these to manage a jitter buffer.

In my application there is basically no contention for the line - at least none that I am creating. Just variability caused by the public internet.

Any kicks in the right direction would be appreciated.

...and that AFAIK is your problem. Packet delay variation caused by Internet weather isn't something managed from either end of the connection. Handling can be suggested using QoS. Being able to ship data over a shortest route is one of the reasons for using a content distribution network.

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unSpawn is spot on. You can configure queuing using TC to give preferential treatment to RTP on your system, but the internet is going to take no notice of your QoS policy and for QoS to work all nodes in the transmission path need to support the same policy. unfortunately the internet does not support QoS.

What you can do is increase the jitter buffer on the application itself. If its video streaming it will help but if is bidirectional interactive traffic the increase in latency will very rapidly cause it to become unusable.

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Thanks guys -- I get it.
I appreciate the confirmation that I am not missing anything ... I will work on the application's jitter tolerance.

While basically true there's nuances. IMHO it would be better to say some network devices (routers, firewalls) may honor QoS...

Technically you are correct, but if one happened to it would be just because the default config uses some default queueing method that provides weighting related to QoS markings. e.g. if you installed a Cisco router on a sub 2M link you would get wfq as the default queueing mechanism which weights based on IP precedence. Its not really QoS as in an end to end service guarantee.