3CX calls jumble or getting cut

[Anthony R]

Hi team,

We're currently facing issues with a small account, from time to time, the calls are getting cut ir present jumbles.

Their actual internet provider is Ayera, they stated the following argument for this ti be happening:

The issue appears to be towards the remote party. After I collected packet captures I noticed that the traffic from the IP phones are not adding QOS markings to the VoIP traffic causing it be treated as bulk traffic. Could you look into having the traffic marked with the following DSCP values?

RTP: EF (SDCP 46)
SIP: CS3 (DSCP 24)
Is this something I should configurate directly on the IP phones templates, into the 3CX PBX or a router configuration? I ran a Monitor Connection Quality and I got the following results:



Should I open a support ticket to get this solved or there is something I could do to sort this out?

Thanks for all your support.

Sincerely,
Anthony

 

[KyriacosS_3CX]

Hi there,

Could you give some more detail on the setup? At first glance this would indicate that perhaps a change is needed on your router and switch to do the QOS marking centrally, or if there is an ALG look into disabling it, although I assume you have a Green firewall check?
https://www.3cx.com/docs/firewall-checker/#h.ikh44hto4r56

Best,
K.

 

[Anthony R]

Hi there,

Well the main issue is that the call quality isn't good, there are pieces where the communication gets cut or starts to jumble.

I ran the firewall test and everything went fine, SIP ALG isn't detected.

We tried changing the router phone just to discard a misfunction but the issue persist. We also added an external extension and it worked fine, so definitely is something that has to be configured inside their LAN to improve the QOS, so these changes should be done directly on the router configuration I guess.

Could the IP phone also be part of the issue? They are some not brand new Polycom VVX411.

 

[KyriacosS_3CX]

Hi,

You are not mentioning the setup you have. If you are self hosting do a packet capture and use wireshark to check where the bad audio originates from.

https://www.3cx.com/docs/capture-network-traffic/

Best,
K.

 

[ivank]

According to the monitor stats above, there are no issues from the phone side.
The RTCP stats from the SIP trunk do not come back to PBX, so it is impossible to estimate the leg PBX --> trunk.

>> Is this something I should configurate directly on the IP phones templates, into the 3CX PBX or a router configuration?
Phone templates won't help.

If possible, your best option is to assign the corresponding DSCP value to the external RTP range in your router.
But this will work only if your PBX is on premise.

At PBX level this can be set manually until PBX config is changed for any reason.
You may try opening a ticket and ask to set TOS value to 46 for your instance.