Changing sample rate (p-time/payload) on SIP Trunk

[carmen.leroux]

We have been battling with bad voice quality on a dedicated voice ISP link. My ISP is ECN.

Upon analysis of the RTP captures they see that RTP packets from the 3CX server are delivered to the ECN network at an inter arrival rate of 200ms every 10 packets sent, this already creates challenges when it comes to transmitting voice packets.

They have suggested that we advertise the packet sample rate(p-time/payload) at which we would like to send RTP on our SIP Invite to 20ms.

As far as I know there are no options to change these settings. Can you please confirm my suspicions?

 

[YiannisH_3CX]

HI @carmen.leroux

I can confirm your suspicions. There is no way to set this option however the default ptime followed by 3CX is 20ms. I would recommend also running a capture on the 3CX server and see the RTP packets leaving the server from there to determine where the delay is introduced.
What PBX version are you running?

 

[carmen.leroux]

Hi @YiannisH_3CX
Thank you for the quick response and the confirmation.

I am running 16.0.8.9.

Is there some documentation that states the default ptime used by 3CX which I can send on to my ISP?

 

[YiannisH_3CX]

I am afraid there is not. The only way I know is to run a capture and look at the timestamps of the RTP packets or you could look at the Delta value that wireshark reports for the specific RPT packets. If it constantly hovers around 20ms then your ptime is 20ms. Delta and ptime are not the same things but if you have consistent audio then the values will more or less match. If the ptime followed is 200 then your Delta should be around 200ms.

It should look something like this:

 

[carmen.leroux]

Thanks @YiannisH_3CX
Appreciate the response and information!