Solved Random One-Way audio

[Ronin]

Hello,

We are encountering a one way audio problem at random intervals.
Unfortunately it is not easy to capture a packet that replicates the error so we currently have only 2 packets with 2 calls that describes it.

This is a screenshot of a call that had the one way audio problem:



Because we use a Patton gateway, we have also reported it to them and they advised us to fix 3 issues:

a) Calls that invoke REFER will be terminated, you should not send REFER (call-transfer) to Patton, because it will be rejected by the provider, you should handle it internally.
////// How can we do that? Does this mean that we have to disable the "Support Replaces" option in the sip trunk? ( we have tried it but we lose many calls )

b) When the call connects -- out from the IVR and connect you (3cx ) switch codec to PCMU.
Do not do that -- do not switch codec prefer PCMA at first we are in Europe -- set all your IP phones to PCMA codec first.
////// Don't the yealink phones use the 3cx's codec priority? We have Set PCMA on the top and PCMU is the last option.

c) 3cx sets up rtp between Patton and the phone directly, also do not do that rather proxy all RTP on 3cx if you can do that.
////// How can we achieve that?

The one way audio problem also exists on softphones and deskphones.

Does anyone know how to fix these issues?

Info:
1) 3cx version: 16.0.622
2) pbx is locally hosted (Debian 9)
3) Yealink t42s with custom templates and softphones
4) Patton gateway (SN4960/4E120) //// Firmware: R6.8 2015-11-13 H323 RBS SIP
5) Firewall checker passes

 

[eddv123]

I see no reference to the type of Patton gateway in your "info" section - however you have provided more information than most. Is it BRI/PRI/FXO/FXS and a supported variant: https://www.3cx.com/voip-gateways/ even the firmware version on the box would be useful.

Also in the above callflow I can see (as per your email REFER) is invoking a 403 forbidden repeatedly. Is this issue only occurring during call transfer also ? I would assume yes ? since you have not posted a standard callflow as far as I can see.

Don't the yealink phones use the 3cx's codec priority

Should be G711 alaw first if settings are correct on 3CX - I have just checked my handset and this is set as so - I am in England.

3cx sets up rtp between Patton and the phone directly, also do not do that rather proxy all RTP on 3cx if you can do that.

I think you are referring to "PBX delivers audio" which will proxy audio via the PBX rather than between endpoints directly, this can be done per endpoint/extension but you probably may also notice that this setting exists on the trunk also, as per: https://www.3cx.com/docs/sip-trunk-options/

 

[Ronin]

Thank you for your answer!

I see no reference to the type of Patton gateway in your "info" section - however you have provided more information than most. Is it BRI/PRI/FXO/FXS and a supported variant: https://www.3cx.com/voip-gateways/ even the firmware version on the box would be useful.

I have edited the 1st post: Patton gateway (SN4960/4E120) //// Firmware: R6.8 2015-11-13 H323 RBS SIP

Also in the above callflow I can see (as per your email REFER) is invoking a 403 forbidden repeatedly. Is this issue only occurring during call transfer also ? I would assume yes ? since you have not posted a standard callflow as far as I can see.

We haven't had reports that it happens in call transferring but it could be.
As you can see in the screenshot the codec changes midway from 711A to 711U. Is this a normal behavior?

I think you are referring to "PBX delivers audio" which will proxy audio via the PBX rather than between endpoints directly, this can be done per endpoint/extension but you probably may also notice that this setting exists on the trunk also, as per: https://www.3cx.com/docs/sip-trunk-options/

Ok, we have enabled PBX delivers audio in the sip trunk settings and also in the extensions - options - troubleshooting and we monitor the situation.

 

[JohnS_3CX]

Just to add:

In all the phones, ensure (G711/PCM) A-variant is on the top.
Do the same for the Gateway
Do the same for Settings > General > Codec Priorities


3CX will prioritize the top codec of whoever made the call (very broadly speaking) so you might want to remove PCMU G711U from any codec list (PBX/Phones/Gateway)

 

[eddv123]

The SN4960 is a fairly old unit however should still work OK, can you confirm if you are using a standard template with the unit. The latest firmware for the unit is SmartWare 6.11 so I would advise on top of what Patton have said downloading and updating the firmware on the unit and if not already refreshed the template to the latest version (must use 3CX supported template).
https://www.patton.com/support/upgrades/

Failing that logging into the unit and running the following set of commands whilst replicating the call will also give you information on what is going on, on the Patton side:

enable
show running‐config
debug call‐router
debug call‐control
debug ccisdn signaling
debug context sip‐gateway signaling detail 5
debug context sip‐gateway transport detail 5
debug context sip‐gateway error

Then "no debug all" after to cease the log flow.

These may differ slightly between the Smartware and Trinity operating systems so use this as reference: https://www.patton.com/support/kb_art.asp?art=446&

Although this is for Trinity based smartnodes commandline, this video gives the jist of the process:

 

[Ronin]

Thank you for your answers!

The problem was solved by
1) Disabling the: "Supports Re-Invite".
2) Enabling PBX delivers audio.
3) Setting the PCMA on the top of the codec list (phones, gateway, 3cx codec priorities)