Solved no sound between (currently) 1 phone and other phones on another site

[GVB]

Hello,

While the implementation goes on the understanding of 3CX/SIP increases aswell.

But there's one thing I can't seem to figure out.

When I call from a (Fanvil 3XG) phone on site A to another Fanvil phone on site B everything seems to work exept for the sound.

When I call with another phone (Snom handheld) on site A to the same phone on site B it works fine.
(this is over ipsec VPNs by the way)

A wireshark capture only revealed a SIP/2.0 407 Proxy Authentication Required for my phone to the 3CX server in the middle of a lot of SIP packets.
Which is strange because it is registered and can call to other phones without problems.

Is there something else I should look at?

 

[eddv123]

Disregard the message you are seeing - this is related to SIP signalling anyway - your issue is audio related. If you wanted to view the audio path in Wireshark however this would be a viable option.

However if you to the extension/extensions you are using in 3CX and enable the setting "PBX delivers audio" - I would say there was a good chance this would solve this.

 

[GVB]

yes, that solves it

what could be the cause of the issue?

and what are the consequences of this?

we have 3CX currently installed on a Gigabyte Brix mini pc.

would this mean the 3CX server would use a lot more resources since the audio is all passing through it instead of just going between the phones?

 

[eddv123]

In a standard generic SIP call the SIP signalling and SDP is setup via the PBX/Platform until the point of call establishment. After the call establishes (and you talk) RTP media is then passed between the 2 endpoints (not via the PBX itself). SIP will then take back over when the call is ended.

What you are doing with this setting is passing the media streams via the PBX - with 3CX this is required when using remote phones via the STUN method, call recording and I myself have had to use this when deploying with VPN.

I have not seen it documented 100% whether this is a requirement as the endpoint itself in a VPN environment should simply be seen as a LAN > LAN device in another subnet - I have always used this setting however.

If you look at the network guide for VPN here you can see that the audio path is going endpoint to endpoint (not via the PBX): https://www.3cx.com/blog/docs/network-configurations-supported-3cx-phone-system/

It will add a little extra resource to the system however I doubt it would be enough to make any major difference.

 

[GVB]

ok, still strange that it works fine with another phone tho.

Thanks for the hint.

 

[eddv123]

If you want to delve deeper possibly compare PCAP/Wireshark traces from both phones and see how the audio path is travelling. See "playing audio" here: https://wiki.wireshark.org/VoIP_calls

What I can say about my experience with snom phones with 3CX is that I have found them a much more solid option compared to some of the other supported phone models.

 

[YiannisH_3CX]

When a call is established the PBX will check if the extensions can directly talk to each other and if they are both local the PBX will let them know that they can talk to each other directly.
In your case since you are using a VPN both the extensions appear as local and theoretically could talk directly to each other if the VPN is setup correctly. If audio does not get through then the option "PBX delivers audio" can help you deal with this as the PBX will handle the audio routing.
To see why the Fanvil device fails where the Snom device works we would have to know the network topology and the setup of the PBX. Perhaps the Snom device is in a different Vlan or it has recordings enabled which would affect audio routing.

 

[eddv123]

Glad to get clarity on the VPN question - thanks.

Perhaps the Snom device is in a different Vlan or it has recordings enabled which would affect audio routing.

FYI it is the Fanvil device that is causing the issue not the snom. If call recording was enabled then the audio stream would go via the PBX.

 

[GVB]

I had some missery with NAT happening over the VPN and packets arriving at the wrong server (it's a thread from a few days ago).

I got it working by using an extra source NAT rule without masquerade for packets where the source is my remote subnet and the port is 5060.

server is 192.168.0.52
this phone is 192.168.0.155
and the remote site phone has a 192.168.30.x address

I attached what I see in wireshark on the local extention that dialed the other site's extention.

I don't know what other info you need?

 

[GVB]

I removed the port number from the source nat rule and now I can call with sound in both directions.

Strange that the SNOM had no issues with that.

 

[YiannisH_3CX]

Glad to see you were able to solve the problem.