Packets are sent in bursts, so increasing the burst size from 3 to 10 packets of 1500 octets gives you a burst of 15000 instead of 4500 octets.
This isn't quite true. Burstyness is an indication that the connection is being underutilised due to the sliding window size being too small. Ideally, the window should not completely fill and the transmission should therefore happen in a non-bursty fasion at close to the link speed.
At the high level statistical point of view, you are absolutely correct - this is the "steady state" TCP was designed for. However, modern NICs use Interrupt Coalesce and similar techniques to reduce the CPU load, and this causes burstiness when looking at the traffic at the microscopic level.
on modern networks, the network speed is relatively high
Not always true. Some of our resellers even have customers running VoIP on satellite links.
However, ISP's don't use QoS to prioritize RTP traffic on your public internet ADSL connection.
You're using the wrong ISP then. [...] QoS over ADSL lines is quite common [...]
We have resellers in many countries, and sure you can buy QoS as a value added service on your MPLS line from any decent telco, but none of the ISPs I have ever heard of provide QoS on ordinary public internet ADSL lines. If you can point me towards just one ISP that advertises DiffServ or similar tag based QoS on their public internet ADSL lines, I will be happy to correct my statement from "ISP's don't use QoS to prioritize RTP traffic..." to "Most ISP's don't use QoS to prioritize RTP traffic...".
In traditional HTTP, the browser opens a large number of connections at once, which, if you aggregate the traffic flow from these connections, gives an effective increase in the initial window size anyway.
Agreed!
Secondly, the very design of the internet precludes latency sensitive applications from working reliably over (comparatively) low bandwidth connections unless QoS is employed to allow that traffic to jump the queue.
Mostly true, except there are alternatives to classic QoS (i.e. tag based QoS) that do allow you to use latency sensitive applications over (comparatively) low bandwidth connections. But if protocol designers at Google (and elsewhere) work under the assumption you just described (roughly paraphrased: "latency sensitive applications do not belong in non-QoS networks") we can all wave goodbye to Skype, GoToMeeting, WebEx and a lot of other realtime collaboration tools that use the plain internet as transport medium! I originally commented this story to warn against turning the most popular protocol on the internet (HTTP) in that direction.
(I'm using "comparatively low bandwidth" loosely here - the important measure is actually how long the transmit queue is on the router compared to the link bandwidth - i.e. how long it will take to flush the queue over the link. This is the important measure of jitter, not things like initial congestion window settings).
Yes, I absolutely and completely agree that jitter and latency is key to the user experience!