Set minPlayoutDelay for WebRTC

[tantonj]

I'm having an issue right now with my JsSIP implementation. I setup free switch and a turn server and I am able to talk with intercom devices through JsSIP. I figured out how to use the icecandidate event and call e.ready() to start the call immediately after finding all candidates.

However, when the call connects there is a small delay in the audio, which grows significantly (over 20 seconds delay). If the call stays open for 20 minutes the delay goes back to 2-3 seconds and then continues to climb again.

Searching for this problem has brought my attention to the MinPlayoutDelay and MaxPlayoutDelay properties of WebRTC. I'm wondering if there is a way to set these in JsSIP. If there is, will it even solve my problem? Has anyone else come across this issue before? I figure this is either something that can be fixed on WebRTC side or my turn server.

A little more information about my infrastructure:
I have an annoying requirement to use windows for everything (no linux allowed). So I am using a version of freeswitch for windows. My intercom devices are the 2N IP Versos. Using my freeswitch server, these 2 devices can call each other directly with no delay. I am using a windows turn server that can be found here. It is the only one I could find that worked to allow local only traffic. Using wireshark I have monitored stun and rtp packets on server, the webclient's machine, and I have downloaded pcaps from the intercoms. In all cases I see no dropped packets.

[tantonj]

The solution here was freeswitch side. In my freeswitch config sip_profiles/external.xml and sip_profiles/internal.xml I changed the tag

<param name="rtp-timer-name" value="soft"/>

to

<param name="rtp-timer-name" value="none"/>

This eliminated the outgoing lag from JsSIP.