we have problems with the sound quality during GotoMeetings.
It has dropouts.
I just installed Goto Network Test but don't understand how to interprete the results.
It's about the bandwith.
It tells about concurrent calls in the headline but about simultaneous meetings in the first column.
So what ?
What are concurrent calls ? one call = one participant in the meeting ?
Or one call = one meeting ?
We don't have simultaneous meetings, we don't provide a server. We are participants in a GotoMeeting.
And the necessary bandwith - is it upstream, downstream or both ?
And for what is the required bandwith intended ?
One participant ? All participants ? A server ?
Thanks for any help.
@berndlentes That chart represents the range of bandwidth demands, based upon the number of users on the same network accessing GoToMeeting simultaneously + the type of information being shared. You would only look at the first line if you are alone on your network.
Regarding audio drops, in my experience this is most often caused by latency within the LAN itself (assuming your ISP upload speeds are 5Mb/s or higher). I would try running a trace route to 'egwglobal.gotomeeting.com', looking for any times returning anything over 100 ms.
thanks for your answer. That's very helpful. I tested with netstat and saw the the GTM software has connections to about 10 different hosts, located all around the world (Amazon Cloud, Frankfurt ..)
Is only 'egwglobal.gotomeeting.com' the important one ? We are located in Germany, isn't there a server to connect us in Germany or Europe ?
Yes, depending upon where you are located we may route your traffic to a closer server. The entire ranges with locations can be found in this support document: https://support.goto.com/meeting/help/optimal-firewall-configuration-g2m060010
thanks for this very detailed information. Which are the domains or IP ranges which deliver audio, video and screen sharing for the conference participants ?
today we had a conference and i made a lot of measurements.
The result of gotonetworktest surprises me.
The upload is very small.
We have meetings with 20 participants. That would mean we need a bandwith of at least 20 Mbit/s.
So the upload we measured seems to be a bottleneck.
How is the ratio between bandwith needed by the upload and needed by the donwload ?
Upload is only needed when sharing content such as audio, video, or activities.
Download speeds can be multiplied by the number of joined users on a single network, as you may have figured out already. So 20 attendees would probably be fine with 20 MB/s.
In my experience, VoIP audio issues can often occur through LAN latency itself -- not necessarily the ISP connection speeds once you leave your local network. The trace route I mentioned will show you the times or lost data points at each stop from the endpoint through your network, and into the internet, ending at LMI servers. Times over 100 ms would certainly indicate some problems if you're noticing audio drops.