Just wondering what the jitter buffer sizes are for the sangoma phones when set to low, medium high? Are they somewhere in thrbrsnge of 10ms? 25ms? 50ms?
How aggressive is the adaptive setting?
I have a pretty solid link with low latency to my SIP trunk provider so I’m just trying to tweak the latency on calls to the best possible.
Jitter buffers don’t change latency,
Likewise ping doesn’t meaaure jitter.
You can’t tweak out network impairment either.
Unless you have a special situation factory defaults for phones and Asterisk are fine for 99.999% of users.
The jitter buffer is a queue for packets in case some have been received out of order or delayed is it not? The packets would then be assembled into the proper sequence for playback. The larger the jitter buffer the longer the delay before audio is heard correct?
Once full rhe jitter buffer adds no appreciable latency.
I noticed jitter buffer minimum size on my Polycom phones. They can go as low as 10ms with the default being I think 30ms.
Changing the pTime increases bandwidth slightly (noticed when capturing traffic between my endpoints and FreePBX) but can also shave off another 10ms. The default on both my S500 and VVX300 was 20ms. I also forced g722:10 and g711:10 in asterisk.
I just like tinkering with ways to shave off 10-20ms where I can. Its cool when the cumulative effects of all the different settings make the latency audibly less. If tinkering with settings like the ones I mentioned above results in no effect or poor audio I just revert.
Ya I would never tinker with that as once you have a small hiccup in latency the calls will go to crap and again the latency is somewhat masqueraded since you can only see from you to the SIP Trunk provider and not beyond. You are just looking for lots of issues long term playing with those.