atcomsystems.ca/forum Forums TDM Business Telephone Systems NEC Telephones & Systems Choppy calls SV9100

Hello, using the pfsense, there is no UDP flood protection on as far as I know if wanted those types of blocking would have to be turned on. I look over multiple wireshark captures from the lan side and wan side I see nothing crazy. RTP data looks super clean.
I'm going to clean up the pfsense yes it has NAT on to public IP to the phone server. I don't know why. I'm going to get rid of that and have PFsense just unblock necessary traffic in its firewall rules only and everything else settings wise its got for messing up VOIP/SIP traffic I will turn off or set to the max for making the connections better. I think that will be the big helper for this issue.
The issue is once again at some random time in an outgoing or incoming call locally you will get a pretty annoying "drop"/mismatch of packets = to the outside caller is hard to understand for about 10 seconds then it comes back. You would think the call is about to drop, but it doesn't. I guess they were also saying they can put the outside caller on hold for a second and then bring them back and it clears it up. It screams out of order packets, but wire shark does think it is.
Heres just one of the rtp analysis for one 13 minute call. Another sad thing is I really can't get much traffic to spy on it's so small of call volume.
SSRC
0x38e8ac40
Max Delta
25.70 ms @ 86
Max Jitter
0.74 ms
Mean Jitter
0.11 ms
Max Skew
-4.96 ms
RTP Packets
509
Expected
509
Lost
0 (0.00 %)
Seq Errs
0
Start at
2.086294 s @ 4
Duration
10.16 s
Clock Drift
-34 ms
Freq Drift
7973 Hz (-0.34 %)

Reverse

SSRC
0x856f8885
Max Delta
20.80 ms @ 44998
Max Jitter
0.18 ms
Mean Jitter
0.06 ms
Max Skew
-8.40 ms
RTP Packets
39963
Expected
39963
Lost
0 (0.00 %)
Seq Errs
0
Start at
2.103154 s @ 5
Duration
799.25 s
Clock Drift
-8 ms
Freq Drift
8000 Hz (-0.00 %)

You know, are you getting "jumbo" packets by any chance?

Not set on by default on the switch so i would think they stop there. Also if I take out NAT on the router, I would just turn off 10-29-21 and that's it correct?

You know you should consider getting a free sip line from another carrier just to test baselines. This could be entirely them.

Hmm, that's true.

From what I understand they have tried that in the past not too sure on that. I still will try after I get the routing correct in the firewall.

Ya, seriously without an MPLS or QOS connect you could be testing this all day and it's beyond your control. The only thing you can do that may help is going to 7.29 compression even then you have no QOS from point to point. I run SIP all the time from carriers and like hardly zero issues as long as I have a QOS or some type.

Hello, I totally forgot that I did end up figuring this out fix a stupid easy fix apparently at some point in time the IP card on the phone system had been set to 5mp half duplex got it to switch to 1gb full and all was well. I found this by looking at the managed switch showing status of connection and seen it before I originally posted but must had thought it was the way it was supposed to be. Thank you for everyone's help