[Starlink] 300ms Telecommunication Latency and FTL Communication

[Sebastian Moeller]

Hi Colin,

Thank you very much, also for the paper and analysis you sent after this.


> On 6. Jun 2024, at 15:43, Colin_Higbie <CHigbie1 at Higbie.name> wrote:
> 
> Sebastion, I was not providing any knowledge or data on acceptable latency for video calling. That is not my area of expertise (closest facet of my business merely involves web site responsiveness and start time for playing audio after buffering, both of which are much less sensitive to latency). I can state that, as a user, I would find 150ms measured ISP latency high, not intolerable, but noticeable -

[SM] I agree, and there is the rub, unlike capacity where we often see hard limits like a VoIP call takes ~100Kbps, if link capacity is sufficiently smaller than 100 Kbps VoIP will not work at all (so there will be hard qualitative thresholds), with latency we often see slow degradation of quality... if e.g. VoIP works well at 100ms, it likely will still be virtually identical at 101ms and still reasonably similar at say 150ms... much harder to turn that into a convincing threshold...

> video conferencing is more sensitive to latency than pure voice (in my personal opinion, no study I've read on this specifically), because we watch people's faces for reactions to what we say as we're speaking.

[SM] I am happy to believe you on this, but ti turn this into something useful for my purpose I will need to find something published, preferably peer reviewed. But thanks to the pointer which should help in my search.


> If there is a noticeable lag there, it disrupts the conversation. On the other hand, the same lag in a pure voice discussion, which is inherently less synchronous, would not be noticeable.

[SM] Not sure I fully agree here, assuming video and audio arrive both with the same delay I would guess both suffer similarly from the delay... my gut feeling is as long as natural speech sequence stays intact, that is no unintended collisions due to both speaking at the same time, audio-only and audio-video should both be sort of OK...

> 
> In my prior post, I was using the 150/300 ms figure you provided and saying that IF that's the max acceptable figure for network latency, THEN that's already a problem to only hit that as the ISP because each participant also adds their distance and network delays. For those that are just as quick, that may be fine. However, assuming there's some form of bell curve distribution on latency, many of these will be longer, and some much longer than what your ISP provides to their customers. Therefore, to ensure a satisfactory experience with the majority of prospective video call participants on other networks, the ISP would need to provide a sufficiently low latency to accommodate these differences. Otherwise, a significant portion of the calls would be of poor quality. Obviously, they can't make up for a participant whose own latency exceeds 300ms, but they should not be the cause of poor communication with someone at 160ms latency. But that's just reasoning around your numbers, not data.

[SM] I am looking at the studies underlaying ITU-114 delay quality assessments, but I think that is a long shot in convincing the regulsator, these numbers are deeply entrenched by know, so I likely would need a more recebt study that conclusively shows that these numbers are way too high...


> 
> That said, here are some studies I found that may be helpful:
> 
> This one includes the 300ms round-trip time, but puts at the extreme outer range of acceptability:
> "Defining 'seamlessly connected': user perceptions of operation latency in cross-device interaction"
> https://www.sciencedirect.com/science/article/abs/pii/S1071581923000770
> 
> "What Are Good Latency & Ping Speeds?"
> https://www.pingplotter.com/wisdom/article/is-my-connection-good/
> 
> A Cisco discussion that supports the 300ms round trip time:
> "Acceptable Jitter, Latency and Packet Loss for Audio and Video on a WebEx Meeting"
> https://community.cisco.com/t5/webex-meetings-and-webex-app/acceptable-jitter-latency-and-packet-loss-for-audio-and-video-on/m-p/4301454
> 
> 
> These are behind pay walls or require academic credentials, so don't know if they are good or not, nor what conclusions they reach - they could even be the source of the 150/300ms figure, but I agree with you that seems high: 
> 
> "A Study of the Effects of Network Latency on Visual Task Performance in Video Conferencing"
> https://dl.acm.org/doi/10.1145/3491101.3519678
> https://www.academia.edu/98061737/A_Study_of_the_Effects_of_Network_Latency_on_Visual_Task_Performance_in_Video_Conferencing
> 
> "Effect of latency on social presence in traditional video conference and VR conference: a comparative study"
> https://ieeexplore.ieee.org/document/10402741
> 
> "Determination of the latency effects on surgical performance and the acceptable latency levels in telesurgery using the dV-Trainer((r)) simulator"
> https://pubmed.ncbi.nlm.nih.gov/24671353/

[SM] Thanks a lot, I will go look these up.

Regards
	Sebastian


> 
> 
> -----Original Message-----
> From: Sebastian Moeller <moeller0 at gmx.de> 
> Sent: Thursday, June 6, 2024 3:22 AM
> To: Colin_Higbie <CHigbie1 at Higbie.name>
> Cc: starlink at lists.bufferbloat.net
> Subject: Re: 300ms Telecommunication Latency and FTL Communication
> 
> Hi Colin,
> 
> 
> 
>> On 5. Jun 2024, at 19:58, Colin_Higbie <CHigbie1 at Higbie.name> wrote:
>> 
>> Sebastian,
>> 
>> At 300ms RTT, that would mean the starting point for any communications are already at the threshold of unacceptability.
> 
> [SM] Not according to the ITU (114)
> mouth-ear delay in ms (so OWDs)
> 0-200ms: users very satisfied
> 200-275ms: users satisfied
> 275-375ms: some users dissatisfied
> 375-600: many users dissatisfied
> 600-...: nearly all users dissatisfied
> 
> So even 150ms OWD still falls within the very satisfied range if the remaining delay is not to large... And even if er string two of these users together, we end up with worst case >300ms delay, but that sill only gets us into the "some users dissatisfied" which the regulator might find an acceptable trade-off in the context of guaranteed internet access parameters (where the idea is the 150ms OWD or 300ms RTT is not the target, but the threshold for being acceptable).
> 
> My gut feeling is these ranges are not actually measured in a way they are now interpreted (e.g. when testing transatlantic call delays users likely already had an expectancy of longer delay and simply judges these calls against a different yard stick). BUT unless I can demonstrate that the original studies resulting in these numbers are terminally flawed there is little chance that I can convince our regulator to take my word vor voice delays over the word of the ITU... so I need different, preferably newer data and focus on probably remote desktop usage as a relative novel use case without much encrusted ideas about acceptable latency...
> 
> 
>> I would think the strongest argument is that's at best a passable latency in absolutely perfect conditions, which never exist. "Pleasant" communication latency is sub-100ms, adding additional travel time to the actual servers involved and processing at each end, the ISP needs to do significantly better than that target to provide some margin for those other sources of latency, many controlled by fundamental physics sending the signal over distance.
> 
> [SM] Personally I agree, yet I am not sure picking a fight over the VoIP numbers is going to be productive, as I have considerably less clout with the regulator than the ITU...
> 
> Regards
> Sebastian