As far as Networked... what if you have your house wired with bluetooth speakers and youre having a party or some other get to together. Pull up your laptop/desktop, tell Pulse that the audio output device is the bluetooth speakers, setup a playlist and all your music will be sent throughout the entire house to the speakers seamlessly.
From a user perspective PulseAudio is fantastic. The only thing lacking is the ease of mixing different channels, especially for INPUT.
Example if I want to record audio from a youtube clip playing in Firefox, there's little means to do this graphically. I may be able to do this in a commandline but hey, that's not to be expected from majority of Users.
Your original claim is invalid as Pulse audio system mode is running PA via global single daemon instance and is disabled by default.
As explicitly stated by PulseAudio Developers, if you want to mess with system mode, you should understand how PA works or your will be ignored.
So go understand PulseAudio before you flood, so your can use "critical eye" and not "slimy tongue" as of now.
To record audio from youtube clip, download the clip using any plugin for your browser (at least 3 working), then drag-n-drop the clip onto Audacity 2.0+ instance.
How about PulseAudio recognize the on-board chipset features and actually have it configured for your system, instead of always haven't to bang on it to get it to recognize anything that is Intel Sound via RealTek 889 never mind 892. Each time some modification arrives in Debian my 7.1 turns into a 2.1 sound configuration. It's embarrassing that Audio in 2012 is still a joke in Linux.
Jack is really a hardware based(realtime access) layer that happens to have mixing capabilities.
Also the example I put could also be a live broadcast and you want to record as the audio comes available, so downloading a video is not applicable in this scenario and is a round about way.
Yate softphone is one of the few SIP clients that can work with my employer's goofy Avaya setup. However, it has no built-in functionality for manipulating audio inputs and outputs -- meaning there's no UI for configuring it to work with my USB headset. So without some method of altering the default, Yate sends output to my laptop's speakers and takes input from my laptop's built-in mic.
Enter Veromix and PulseAudio. After installing Yate and during the first phone call, I can use controls in Veromix to change Yate's audio routing. I can send output to and take input from the headset. Fortunately, I only need to do this once after installing, as the configuration is saved. Furthermore, if I want to switch to speakerphone-like behavior, I can route the output and input back to the laptop's hardware.
Agreed, this is suboptimal. Better would be for Yate to incorporate some audio configuration, like just about every other softphone I've tried. But, alas, I can't get any of those to properly perform (or hold) a SIP registration. The problem lies somewhere, I think, in Avaya's SIP implementation. But I'm no SIP expert, so I'm kind of guessing here.