One way audio issue, tried the basics

[Stijn_DC]

Hi all,

I'm having a real headscratcher here maybe someone else could point me in the right direction.

Some specs:
two different 3CX installs, one RPI 4, one windows, same VoIP provider
3CX 16.0.8.9 on both machines
Sitting behind the same make/model of firewall but different sites and IP addresses, identical firewall rules.
both 3CX firewall tests are green (Full Cone NAT + no ALG)
3CX delivers audio on the trunks has been checked, re-invite and replace has been unchecked.
same for all extensions. 3CX delivers audio, re-invite and replace has been disabled.

When making a call, 3CX => external number, we have two way audio.
When taking a call, external => local extension, we have two way audio
When forwarding a call to an extensions voicemail => One way audio, external caller hears the message to record. But the recordings are empty.
When forwarding a call to an external number => One way audio, external caller can hear the person on the line, but calleé that receives the forward hears nothing
when forwarding a call to an 3cx app on mobile => One way audio again, the external caller hears the person on the line but the app is silent.
When making an internal call 3CX remote extension on mobile app to 3CX internal extension => two way audio.


I'm currently leaning towards an issue with the Provider, since the moment a call is being transferred there is no audio.
When tracing this kinds of calls, I can also only see audio leaving the 3CX server, but never receive audio.

Any help would be greatly appreciated.

 

[Alphabetic]

Run a Wireshark trace to see if audio is passing through.

 

[Stijn_DC]

All traces look like this.
I can see audio leaving the 3CX, but I never see it return.

 

[ChrisC_3CX]

@Stijn_DC
Regarding the 3CX Extension settings, do bear in mind that by default, "Supports re-invites and replaces" must be enabled so I do recommend switching it back.

Based on the provided screenshot it does indeed seem that the SIP Provider is not sending any RTP traffic, provided of course that the right hand side is the SIP Provider's media server and the middle is the 3CX PBX. You might want to also try and actually filter the packet capture with the UDP port(this case 13440) instead of just looking at the SIP Call flow just to make sure that traffic is indeed only one way. If this is the case, you may need to contact the SIP Provider.

By the way, is this a 3CX supported SIP Provider? Could you provide the information mentioned here: https://www.3cx.com/community/threads/information-to-provide-when-requesting-help.67558/

 

[Alphabetic]

Is the firewall checker passing? Have you been restrictive in your firewall from your SIP provider?

 

[Stijn_DC]

@ChrisC_3CX You are correct in assuming that the 3CX is in the middle. On the left and right hand side are the IP adresses of the Provider.
I've re-enabled the settings for the test user.
The SIP Provider we are using in both cases is CM.com.

I will take a look at the dump and post feedback ASAP

@kieferschild Firewall checks pass without issues.
Our firewall has been setup to disable ALG on all VoIP platforms (RTP, SRTP, SIP, MGCP, etc.)
our default firewall rules are to allow all 3CX services (5001, 5060, 5061, 5090, 9000-10999) from the PBX out and from the internet in. That means that (for now) it allows any connection on any IP address.
For our NAT rules, i've set it up so that all communication on those ports is not translated.

 

[Stijn_DC]

When changing the tracing settings and fiddling with different parameters, I can only conclude that only UDP traffic on port 13440 is originating from the PBX side to the IP of the provider.


When checking the PCAP file for connections from the IP adres of the provider I also see nothing coming back.

• 3CX Version, Pro 4SC 16.0.8.9
• Server OS, Raspberry Pi & Windows 10
• Is the 3CX Server Hosted and where? Not hosted, local installation
• IP Phone Make/Model/Firmware: 3CX Softphone current version
• Provisioning Method: Local / VPN / STUN / SBC: remote 3CX softphone or local
• Trunk Provider or Gateway Make/Model: CM.Com
• Has the Firewall Checker passed: YES
• Are custom Phone Templates being used: NO

 

[ChrisC_3CX]

@Stijn_DC
Thanks for the update.
Could you also provide me with the packet capture via a PM?

 

[Stijn_DC]

I've sent the over.

 

[ChrisC_3CX]

@Stijn_DC
I've checked all provided packet captures and indeed, there is no RTP traffic coming in from the SIP Provider in none of the cases and for both PBXs. Everything else in the SIP Flow looks normal so I think it would be best to have a talk with the provider as well. Before doing so, just in case, do make sure that you have made any modifications to the providers template in 3CX. You could just create a dummy SIP trunk of the same provider and crosscheck the default settings with the ones you already have configured on your existing SIP Trunks.

 

[Stijn_DC]

we've escalated this case with the team @cm.com, but no word yet, I will keep you posted

 

[ChrisC_3CX]

Good to know! Let us know what happens and if there is anything else we can do to help!