Great! I am greatfull to help community. I am great fan of Windows as I know it best. And also FS likes Windows. I am reading a lot how people have problems with building FS on Linux, but on Windows is matter of seconds. Also there are a lot of good libraries to connect FS ESL with .Net. Anyway, how did I figure it out? Unfortunately there are not a lot of us on Windows to brainstorm about, but leg in audio can be caused only because of timing. And this parameter was invented only because of Linux core. This was my assumption and it worked it out. But I would always suggest using bare metal when go voip and wants to go serious. Best regards, Gregor On Fri, 6 Dec 2019, 20:09 Jaan Kaja, <Jaan.Kaja@enghouse.com> wrote:
Many thanks, Gregor!
It now works like a charm on my bare metal workstation. Audio comes up fast after call setup, and the audio delay buildup is gone. How did you figure it out?
Regarding virtualization, a Windows 2016 Server under Hyper-V still has the problem, but it works fine under VMWare. It may be that people have done more performance tweaking on the VMWare system.
/Jaan
*From:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> *On Behalf Of *Gregor Nanger *Sent:* den 5 december 2019 22:05 *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Long increasing audio delay
Hi!
We are using Windows on production for years, also with verto.
As I remember, we had similar problems. I think we solved with timer settings in verto.conf.xml to:
<param name="timer-name" value="none"/>
Try to set it and see if it works.
Best regards, Gregor
V V čet., 5. dec. 2019 ob 20:10 je oseba Jaan Kaja <Jaan.Kaja@enghouse.com> napisala:
Hello,
We are testing webrtc via mod_verto with the verto_communicator demo. After some initial troubles with certificates, we have it running from Chrome and Firefox. Very nice!
But, when we leave the connection on for some time, the audio in the direction from the browser to a phone becomes increasingly longer. We have clocked ~11 seconds, if we leave it on long enough. The audio in the other direction has a normal IP telephony delay. We have not found any browser dependencies.
The FreeSWITCH version is 1.10.1 (latest stable release). The phone end has been a Blink softphone, or a trunk from a Cisco UCM. The latter negotiated mulaw PCM with a 20 ms packet size.
We have tried different server OS:es:
· Windows Server 2019 gives the fastest buildup of delay.
· Debian installed in a Windows Substrate for Linux gives a slower buildup of delay, but it does continuously increase.
· CentOS virtualized in VMWare doesn’t have any buildup of delay, and the delay is a normal IP telephony delay.
· So, the more Windows, the worse…
I have a Wireshark trace, and I can get a TLS stream from it. I can install a key file to decode the stream, but the configuration wants a protocol, and I don’t know what to enter there. There are no dissectors for Websocket. So I’m at a bit of a loss, as to how to debug this. I don’t see anything obvious in the freeswitch.log.
Questions:
· Has anyone used webrtc on Windows?
· Has anyone had the same problem?
· What can be done to debug the problem? Turn on some logging?
We could package a Linux machine to our customers, but as everything else is on Windows, it would be nice to make it work.
Best regards,
Jaan
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*Gregor Nanger*
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