Wow for some reason KVR stopped sending notifications about this thread... It looks like I am late to the party!! Please find my answers and comments first. Will post more info about my own experience in another post.
What UPnP does is simply open ports on a router, which is required when connecting from the outside (WAN port on the router). It does not change anything to the latency, but if both computers are on the same LAN, this should definitely not be necessary. Are both computers on the same side of the router?UpNp Redirect: No matter the router (I've tried 3 now) I always have to use the uPnP redirect to establish communication between the sending and receiver plugins on the two computers. Is that adding some latency that I can avoid? I am unclear if this should be necessary or desirable. Can you shed more light on this?
Drop outs will actually sound quite different from bypassed plug-in. Also, what matters is audio dropouts count on the receiver side> as long as it remains to 0 no packet has been lost.One unfortunate aspect of the the demo version of the software is that it deliberately inserts silence as part of the demo restrictions. That makes it difficult for someone to know if the software is working well on their machines before buying it.
The purpose of Connector and software-based network audio is to avoid the need for dedicated switches, routers and audio interfaces used for Dante or AVB. "standard" switches should do the job pretty well.The only reasonable option in that list is the FAIR or GOOD category. TSN Nics and Swiches and Ultra Low Latency Nics and Switches are very expensive. If we want audio over ethernet and have that kind of budget then we're looking at Dante.
That's old school (and painful to configure with static IPs), but it may indeed help diagnose any network infrastructure issue.Yes, try a crossover cable!
That's getting better, but it's still pretty high latency for a local network! That's the typical latency I would get for an Internet fiber connection on a single audio stream.Summary: In short it is now reliable for transferring 4 audio streams with lossless audio at 44.1K and 48K without dropouts at both 1024 and 768 buffers on the receiving connectors.
Have these tweaks made any difference? After 5 years using Connector on many platforms/networks, I usually found not much difference while tweaking network cards settings. Connector forces the network card to push packets as soon as they are ready (to avoid extra buffering that causes tons of jitter), so increasing the frame size should just increase the bitrate (frames are stuffed with empty data if the audio data is smaller).First the tweaks that were available to me:
1. Flow Control Disabled
2. Interrupt Disrupt Disabled.
3. Priority and VLan Disabled
4. Speed and Duplex 1.0 GB Full Duplex (was set to auto negotiation)
5. The Receive and Transport Buffers were already maxed at 512 and 128 respectively.
Faster switches/router indeed usually have lower jitter as they are expected to be able to operate faster. Latency on a LAN is not the issue, it's jitter that forces you to increase the buffer size.From what I understand, speed helps with latency--especially there are QoS settings and if jumbo sized frames are sent. We'll never need this much data throughput, but it should help with the over all latency.
Statistics: Posted by Blue Cat Audio — Tue Jan 21, 2025 8:31 am
