WebRTC Audio Jitter Buffer

[VoIP Lurker]

I'm working for a VoIP provider and we're using WebRTC / Verto to connect users to our Freeswitch PBX. I'm trying to figure out Jitter Buffers in WebRTC but I can't find parameters to change it or  documentation about the default values.

Does any1 know?

Thanks in advance!

[David P]

After answering and before entering any conference, try doing this in your Freeswitch dialplan:

<action application="set" data="jitterbuffer_msec=5p:100p"/>

Note that I received this as a tip myself months ago and I can't explain how it works.

[Lars Undercover]

Thanks for your reply David. I'm familiar with this settings. It is documented here: https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/16353845

However this only controls the serverside buffer. I'm looking for a way to change the clients parameter.

[David P]

On Friday, May 10, 2019 at 7:42:00 PM UTC+12, Lars Undercover wrote:

Thanks for your reply David. I'm familiar with this settings. It is documented here: https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/16353845

However this only controls the serverside buffer. I'm looking for a way to change the clients parameter.

Someone pointed me to https://webrtc.org/experiments/rtp-hdrext/playout-delay/