After 15.5 update 6, phone systems lose all audio

[nsymphony]

After the recent .6 update on a couple of phone systems, I'm having issues where no audio on either end of the call is happening. Regardless if it's extension to extension a desk apart, or an external call, it's just dead air. The only way I have found to 'fix' the problem is to reboot the server, restarting services will not fix it. I've checked the event log and there isn't anything showing when the issue occurs. Additionally, prior to the .6 update everything was rock solid. Is anyone else having this issue? The exact version we are are seeing this on is 15.5.15502.6. It's happening on two separate systems thus far...

 

[VoIPTools]

You need to increase your RTP ports. SP6 increased the number of ports used.

https://www.3cx.com/docs/ports/

UDP: 9000-10999.

 

[YiannisH_3CX]

Also in case you meed to allow local ports as well the local port range has been increased to 7000-8499.

 

[nsymphony]

I will give this a try and let you know if it doesn't solve the issue. Thank you!

 

[Kristof Beckers]

it would be really nice from 3cx to have mentioned this not so little change in the change log from the update. But the modified port list did the trick for me.

 

[YiannisH_3CX]

it would be really nice from 3cx to have mentioned this not so little change in the change log from the update. But the modified port list did the trick for me.

Please note that the port increase has been now added to the change log. Glad to see the issue has been resolved.

 

[craigreilly]

Unless your network restricts internal communications, forget about the internal ports.
Router changes:
9000-10999 needs to be 1:1 forward from external to internal.
5060 needs 1:1 forward and should be restricted to STUN clients locations and Providers
5090 needs 1:1 forward for cellphone clients and SBC - no IP restriction
There are a few more ports... such as 443 (or 5001) to publish Presence information and Web Management.

UDP - 48000-65535 needs to be opened Outbound for Webmeeting only.

https://www.3cx.com/docs/ports//

 

[craigreilly]

I need to be able to understand 3CX requirements in multiple network scenarios including multi-VLAN/internal firewall. So the question remains: what are ports 7000-8499 and 40000+ used for and when/where do they need to be opened?

7000-8499 - internal only. I have 2 sites. 1 PBX at site A.
I do nothing with this port range on my router. My 3cx server and phones and all workstations in Site A are on the same VLAN.

40000+ as far as I know outbound only - needed for Webmeeting. I do not restrict outbound traffic, so again, I do nothing with this port range.

 

[YiannisH_3CX]

Hello @mcbsys

Please note that External ports are the ports the PBX uses to communicate with external entities like providers from example. In case of an Invite from a provider the PBX will request Audio on a port included in the external range.
Internal ports are the ports the PBX uses to communicate with internal devices like IP phones gateways etc. When a call is initiated between a local IP phone and the PBX, the PBX will request audio on a port included in the internal range.
However devices communicating with the PBX can request audio on any port as each vendor uses different ports. The 3CX clients uses ports in the 40000 range for example and Fanvil phones use ports in the 10000 range if i remember correctly etc.
When creating rules for ingress and egress traffic you should do so based on the PBX ports as ports requested by devices could be in different ranges.

 

[cobaltit]

So the internal ports were definitely documented on that ports page at some point. It was also shown in 3CX in v12 and v12.5 at least. I'll check in v14 if I still have a customer on that.

 

[cobaltit]

Btw I read that as you never saw the internal port ranges documented. If you mean the recent change then you are correct as I don't see it documented anywhere either. There was a thread where someone put some port down in the 7000 range (maybe SIP) and one of the updates that expanded the internal RTP range clobbered it.

 

[cobaltit]

@mcbsys That's from a v12.5 install I have access to. I tried to find the old ports page in the way back machine because I remember it being listed there at one point but they changed the URL (3CX) so I can seem to track it down. And I want to say it was also in one of the tests at some point. But it does seem to have evaporated from recent documentation for sure.

 

[cobaltit]

Oh yeah, the port range is defined in the parameters section as well.

 

[YiannisH_3CX]

@YiannisH_3CX, thank you for the explanation of internal audio/media ports. Now, what about the other internal ports, 7000-8499? How/where are those used? See my original question in post #7 above.

Internal ports are the ports the PBX uses to communicate with internal devices like IP phones gateways etc. When a call is initiated between a local IP phone and the PBX, the PBX will request audio on a port included in the internal range.

 

[Lee Cramman]

Does this forum have stickies? Because this problem is coming up quite often...

 

[craigreilly]

Does this forum have stickies? Because this problem is coming up quite often...

It is an available feature.

 

[YiannisH_3CX]

Thank you @YiannisH_3CX. I guess I was confused because of the reference to audio on three port ranges. Do I understand correctly now:

7000-8499 The PBX requests that internal devices send audio to the PBX on these ports.
10000+ Fanvil phones request that the PBX send audio to the phones on these ports.
40000+ The 3CX Client requests that the PBX send audio to the client on these ports.

These "requests" are part of the INVITE packet (specifically the Media Descriptor m=audio line) where the side initiating the connection tells the other side where to send the audio stream.

That is correct.

Where is the port range definition for the 3CX Phone client?

Think of the 3CX client as an IP phone. The ports it will request are not part of the PBX but of the client. That is why i recommended creating rules based on the PBX ports not the clients. The client will use for local calls a random port for audio from the range 40000 and upwards to 65536

 

[Lee Cramman]

@YiannisH_3CX any chance we can have a sticky on this subject - it keeps coming up over and over.

 

[mdc2]

I am also having the same issue since installing 15.5 Update 6. Our SIP provider is actually dropped into our network behind the firewall, therefore it is local lan to local lan connection and the ports would not be an issue. However, the calls still just stop providing audio. I have restarted the server completely. Seems to clear the problem for a while. But then it is reoccurring randomly several days later. I cannot determine how to replicate the issue and therefore cannot turn on logging to catch it.

 

[cobaltit]

@mdc2

So you start out by same you have the same issue and then state that because of how you get your SIP connection it shouldn't be an issue. But the port range changes are really the only thing that changed in SP6 so it more than likely applies to you. You've stated your SIP provider is local to your PBX but you haven't stated where the endpoints that are experiencing the issues are in relation to the PBX. The expanded range affects both SIP providers AND remote phones. The internal range also expanded but that shouldn't affect anyone unless they routed private networks with ACLs or firewall between the segments.