I'm currently trying to resolve an issue with audio quality for one of our clients - it appears to be jitter related as the client is on a fixed wireless (cellular) connection.
The outgoing audio stream from FS (to the client's phone) seems to be coming through clearly - I assume the jitter buffer on the Yealink handset is doing its job there.
The issue is with the incoming audio stream from the phone - the packet capture shows a lot of 'wrong sequence number' packets and the person on the other end gets very choppy audio from them.
We don't have any jitter buffer settings enabled in FusionPBX - should we be turning this on to account for the jitter from the client's phone into FS?
I've read that having a jitter buffer on both sides of a connection is not good for audio quality, but then how do we compensate for the jitter on the RTP packets coming into the PBX?
Cheers,
Ryan
The outgoing audio stream from FS (to the client's phone) seems to be coming through clearly - I assume the jitter buffer on the Yealink handset is doing its job there.
The issue is with the incoming audio stream from the phone - the packet capture shows a lot of 'wrong sequence number' packets and the person on the other end gets very choppy audio from them.
We don't have any jitter buffer settings enabled in FusionPBX - should we be turning this on to account for the jitter from the client's phone into FS?
I've read that having a jitter buffer on both sides of a connection is not good for audio quality, but then how do we compensate for the jitter on the RTP packets coming into the PBX?
Cheers,
Ryan