Audio problem on some call (not all call have problems)

Hello, I finished installing my FreePBX server recently, connected with the trunk provider, SIP manager (a gigaset N610 IP pro) and with my odoo. Because of my swisscom router using the port 5060 for his own thing, I was obliged to put my FreePBX server in DMZ (exposed host).

To test if everything work, I have called myself from odoo to my cellphone and from a voip phone (also a gigaset) to my cellphone, everything work fine. I tested to call with my cellphone and receiving the call with odoo or ith the gigaset worked fine too.

But I have found out that for some call, their was no audio and I don’t know why.

Hi again @AubeMort
I don’t have experiance with Swisscom Router but i guess there must have SIP ALG option Enable? Could you pls check that option and Turn it to Disable (OFF).

It is the funny thing, there are no option to disable the SIP ALG, but that is not my problems, I have some calls that don’t have Audio.

Hold on, This logs showing WS/WSS ? this is totally different issue. I thought your SIP Calls via Sip Trunk has RTP issue ?

The wss work fine, I think.
Just that sometimes I have Audio problems like no sound from both side.
But not all call have problems, some the audio work fine.

Why you are not using default PJ_Sip 5060 protocol? Have you tried at all ?
WebSocket (WS) is a bit complicate. WS/WSS required a bit more knowledge for Sip.

My suggestion is try to use Sip Trunk with PjSip 5060. some basic call. After that you can do with TLS or WSS what you wanted.

I don’t think works fine. if you have DEAD Air.. this is not fine.

I am obliged to use wss with odoo, and if by DEAD Air you talk about the closed connection it should be because of the page getting frozen by the browser when inactive (you changed page) or the computer entering sleep.
Like I said my problem is that for some unknown reason some calls don’t have Audio.

After some test, I have at least one person from Sunrise operator for who the audio don’t work from both direction.

I think I need a G711 codec, but I can’t find it.
Source : https://community.sunrise.ch/d/2031-hd-voice-voip/4

G711 or G722 ? confused.

  • G.711 u-law (mu-law): Used mainly in North America and Japan.
  • G.711 A-law: Used mainly in Europe and the rest of the world.

I am in Switzerland, based on the forum linked before, the codec G722 could be blocked by sunrise, I have ulaw, alaw and G711 (and some other) enable in the pjsip setting for my sip trunk with swisscom.
Like I have said, I am sure that at least some call to mobile phone with Sunrise as operator don’t have audio in both direction (the call is connected).

ulaw (µ-Law) and alaw (A-Law) are variants of G.711, so you cannot specify g711, you must specify the specific variant(s).

Specifying too many can cause problems, because systems often can’t cope with packet fragmentation.

I agreed with @david55.
And Also you can try to get some informations from your Provider Sunrise help desk side.

My provider is Swisscom, yesterday I opened a support ticket and I am waiting for update.
Codec I use for my SIP Trunk