One way audio one or twice a day, Intermittent, all other calls are working fine, callback will work fine too


I am having an issue with our IP phone system where it is getting one way audio, the client can’t hear us when we call, if the call is placed right away it will work just fine.

the setup
Freepbx server > VNP > site A
Freepbx server > VNP > site B
Freepbx server > VNP > site C
Freepbx server > VNP > site D
router: Edgerouter POE5

I am at a lost, I have contacted our SIP provider, but they can’t help right away because they need to check with their media peer.

The issue has been happening for a few months as far as I can recall, at first I didn’t think it was our system, but once I started doing recording of the calls it was clear that the issue is with our system and not with bad cell call or something like that on the clients end.

I have provided our SIP provider with tcpdump and wav files already, just waiting now on them to find out what’s going on, been a couple of weeks already.

Anyone experienced this, wonder if this is an issue with my setup or SIP provider? What could be causing one way audio, always happends on outgoing call and client can’t hear.
Please help. thank you

It is a possibility that your rtp port ranges as defined in Asterisk is not copasetic with your router rules or even the VSP’s range.

You could turn rtp debugging on “rtp set debug on” and after a failure find that time in your ‘full’ log ( which might by then be huge :slight_smile: )

I am having the same problem with regards to random 1 way audio. I have TLS and SRTP setup and it works fine 99% of the time.

I’m using Sangoma S500 but struggling to troubleshoot it. If I factory reset the phones they seem to work fine for about half a day and then they start to get random 1 way audio.

Thanks Dicko
I did setup rtp set debug on, but it didn’t help much

Talked to our SIP provider Flowroute, they made some changes, specifically “Our peer tried another route removal”

so far one day we had zero bad calls as far as I know about it.

Seems like the issue was down the SIP provider end.
Well time will tell if it worked. :slight_smile:

Thanks for help!

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