[jitsi-dev] Issues with JitterBuffer when different packetization times are used


#1

Hi all,

I'm using JVB with audio mixing and I'm trying to use Opus and ULAW with different packetization times.

So this is the situation:

  * JVB transcodes audio between Opus and Ulaw;
  * Opus is used with 60ms ptime;
  * Ulaw is used with the default 20ms ptime;

Sometimes the audio is fine, sometimes is choppy and almost not intelligible.

When I have audio issues I see in the log a lot of "Packets dropped because full" in the RTPSourceStream statistics after the call.

My assumption is that the jitter buffer requires some changes to work properly when different ptimes are used.

I tried to disable the adaptive jitter buffer and use a higher value for net.java.sip.communicator.impl.neomedia.RECEIVE_BUFFER_LENGTH but still I have this issue.

Could someone give me some hints to which parts I have to modify to fix this issue? Actually I'm quite lost among FMJ's classes :).

Thanks in advance,

Matteo


#2

After further tests I realised that probably the issue is not in the jitter buffer itself, but the root cause must be somewhere else since with the adaptive JB enabled at default I got the same results.

Cheers,

Matteo

···

________________________________
From: Matteo Campana
Sent: Friday, January 27, 2017 1:44:44 PM
To: Jitsi Developers
Subject: Issues with JitterBuffer when different packetization times are used

Hi all,

I'm using JVB with audio mixing and I'm trying to use Opus and ULAW with different packetization times.

So this is the situation:

  * JVB transcodes audio between Opus and Ulaw;
  * Opus is used with 60ms ptime;
  * Ulaw is used with the default 20ms ptime;

Sometimes the audio is fine, sometimes is choppy and almost not intelligible.

When I have audio issues I see in the log a lot of "Packets dropped because full" in the RTPSourceStream statistics after the call.

My assumption is that the jitter buffer requires some changes to work properly when different ptimes are used.

I tried to disable the adaptive jitter buffer and use a higher value for net.java.sip.communicator.impl.neomedia.RECEIVE_BUFFER_LENGTH but still I have this issue.

Could someone give me some hints to which parts I have to modify to fix this issue? Actually I'm quite lost among FMJ's classes :).

Thanks in advance,

Matteo


#3

Hi Guys,

I guess that no one has ever tried this scenario, am I right?

Cheers

Matteo

···

________________________________
From: Matteo Campana
Sent: Friday, January 27, 2017 3:45 PM
To: Jitsi Developers
Subject: Re: Issues with JitterBuffer when different packetization times are used

After further tests I realised that probably the issue is not in the jitter buffer itself, but the root cause must be somewhere else since with the adaptive JB enabled at default I got the same results.

Cheers,

Matteo

________________________________
From: Matteo Campana
Sent: Friday, January 27, 2017 1:44:44 PM
To: Jitsi Developers
Subject: Issues with JitterBuffer when different packetization times are used

Hi all,

I'm using JVB with audio mixing and I'm trying to use Opus and ULAW with different packetization times.

So this is the situation:

  * JVB transcodes audio between Opus and Ulaw;
  * Opus is used with 60ms ptime;
  * Ulaw is used with the default 20ms ptime;

Sometimes the audio is fine, sometimes is choppy and almost not intelligible.

When I have audio issues I see in the log a lot of "Packets dropped because full" in the RTPSourceStream statistics after the call.

My assumption is that the jitter buffer requires some changes to work properly when different ptimes are used.

I tried to disable the adaptive jitter buffer and use a higher value for net.java.sip.communicator.impl.neomedia.RECEIVE_BUFFER_LENGTH but still I have this issue.

Could someone give me some hints to which parts I have to modify to fix this issue? Actually I'm quite lost among FMJ's classes :).

Thanks in advance,

Matteo