Jitter issues on incoming audio stream

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riza

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Oct 19, 2018
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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.

2020-07-20 14_44_02-Wireshark · RTP Stream Analysis · bell_2.pcap.png

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
 
If downstream is fine and, I assume, the connection is asynchronous? Then the most likely cause is contention (a lack of available) upstream bandwidth. When bandwidth becomes limited, networks will "queue" packets for a short period of time often resulting in the "Wrong sequence number" that you are seeing, rather than a complete packet loss (which does occur in very congested networks).

Can you re-test with all other (non voip) traffic removed from the network?

FreeSWITCH does have a jitter buffer that you can enable from the dialplan, but I would spend some time looking at the network first. Is your router capable of reserving some upstream bandwidth for RTP?
 
Thanks Adrian, we do have QoS / queueing configured on their Mikrotik router, with priority given to VoIP. I will review this setup to make sure it's working as expected.

Do you know the best way to enable the jitter buffer for outbound calls from a specific extension range as a test? Could I just add the below to the dial plan with 'continue' set?

<condition field="sip_from_user" expression="^3\d{2}$">
<action application="set" data="jitterbuffer_msec=60:200:20"/>
 
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