[RTW] Realtime communication over HTTP (Re: Baseline in or out of scope)
Harald Alvestrand
harald at alvestrand.no
Sun Feb 27 00:37:04 CET 2011
(Changing subject again, since this has strayed from the baseline thread)
On 02/26/11 19:47, Bernard Aboba wrote:
> Silvia Pfeiffer said:
>
> "I doubt both of these statements about HTTP are true any longer."
>
> [BA] In fact, they haven't been true for quite a while.
>
> Every day users participate in interactive sessions over
> HTTP, largely in circumstances where use of UDP media is
> not possible. Because of the prevalence of highly restrictive
> enterprise firewalls that do not permit passing of UDP,
> the ability to support realtime communications
> over HTTP is now considered a practical requirement for
> business-oriented services, such as web conferencing.
>
> Although realtime communications over HTTP is largely used
> as a fallback, measurements show surprisingly high
> audio quality in the majority of sessions, probably because
> many sessions take place over well-provisioned enterprise
> networks.
>
The Google Talk numbers I've seen published elsewhere are that ~5-10% of
sessions run over TCP, relayed through a server, because UDP doesn't get
there.
The reasons to prefer point-to-point UDP if possible include:
- Much lower delay when the endpoints are close to each other, network-wise
- Much cheaper provisioning for the service provider
The lower delay is the factor with the largest impact on comfort of
conversation, I think; as long as we don't encounter congestion, the
audio quality shouldn't be that much different.
When we encounter congestion, audio-over-TCP will experience this as
jitter, while audio-over-UDP will experience this as packet loss, so the
experience may be different.
There are many tricks available for lessening the impact of both.
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