Poor performance with WebSocket Streaming when having bad latency

[lidanh]

Hi Giuseppe,

Thank you for your reply.

There is no need to go deeper into the client-side solution, I don't think this is the way we want to go.
In any case, I would still like to test the performance with larger frames per WebSocket packet. The performance were improved significantly since we changed the sendbuf value, but I still wonder if we can improve it even more.

I'll mention that we tried testing this through that open source WebSocket server. We had the best performance by increasing the size of each WebSocket frame (about 10kb each instead of 1.2kb Lighstreamer is doing) in addition to to the sendbuff change behavior.

Thanks,
Lidan.

[giuseppe.corti]

Hi Lidan,

Okay, if you want to go down that road then you have to use <MSS> parameter, which is considered private and that's why you did not find it in the lighstreamer_conf.xml file.
Please add something like

<MSS>10000</MSS>

in lighstreamer_conf.xml, for convenience immediately below the setting of <sendbuf>.

But please note that the factory setting is considered to be the optimal trade-off between the different network scenarios possible. I remind you that it is not excluded that in some cases the TCP nevertheless decides to break up into smaller packets.

We therefore recommend that you perform various tests in your typical scenarios to assess the value of <MSS> or if it is appropriate to return to the default value.

Regards,
Giuseppe

[lidanh]

Hi Giuseppe,

Thank you for your help.

The MSS value seems to be working. The performance of the snapshot were improved as well. We are considering leaving the MSS with the value of 6000.

By the way, is it possible to perform GZip compression when using WebSocket? I looked at the Server configuration and I noticed that it doesn't work with streaming. Is there any other way to get it working with streaming?

Thanks,
Lidan.

[giuseppe.corti]

Hi Lidan,

I can confirm that for the moment compression is only available for HTTP streaming and not with WebSockets.
But we are working on that, waiting for the officiality of draft "Compression Extensions for WebSocket" ( draft-ietf-hybi-permessage-compression ).

[lidanh]

OK,

Thank you Giuseppe!

[HoangTranVinh]

Dear all,
I have the same problem.
I am lucky when I found this thread

I want to know:

If I set the MSS:
<MSS>10000</MSS>

then the value of <sendbuf> should be ?
and other relate parameters (if any)

Thanks for your support,
HoangTV.

[giuseppe.corti]

Hi HoangTV,

If you decide to change the factory default values for the parameters like <sendbuf> and <MSS> then their tuning should be the result of intense tests in your typical scenario.

In any case, I expect that the value of <sendbuf> is always kept higher than that of <MSS> (even slightly, in your case i can speculate 12000).
But, please let me to repeat that you should thoroughly test the choices taken.

Regards,
Giuseppe