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
 

Adrian Fretwell

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Aug 13, 2017
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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?
 

riza

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Oct 19, 2018
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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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