WebPhone - one way audio

Hello!
I’m facing a different problem: My webphone has finally connected, they have managed to communicate but only one side can talk (usually who initiated the call).

This is the CLI log while the call is active:

0x7f7ba46dd480 – Strict RTP learning after remote address set to: 177.129.4.62:42934
– PJSIP/101-000000a5 answered PJSIP/100-000000a4
0x7f7ba46e2a70 – Strict RTP learning after remote address set to: 177.129.4.62:60897
– Channel PJSIP/101-000000a5 joined ‘simple_bridge’ basic-bridge <5f530cc2-438b-400d-b1de-bd3130bd2ba9>
– Channel PJSIP/100-000000a4 joined ‘simple_bridge’ basic-bridge <5f530cc2-438b-400d-b1de-bd3130bd2ba9>
0x7f7ba46dd480 – Strict RTP learning after ICE completion
0x7f7ba46e2a70 – Strict RTP learning after ICE completion
0x7f7ba46dd480 – Strict RTP qualifying stream type: audio
0x7f7ba46e2a70 – Strict RTP switching to RTP target address 192.168.1.8:60897 as source
0x7f7ba46dd480 – Strict RTP switching source address to 192.168.1.8:42934
0x7f7ba46dd480 – Strict RTP learning complete - Locking on source address 192.168.1.8:42934
0x7f7ba46e2a70 – Strict RTP learning complete - Locking on source address 192.168.1.8:60897

(PS: I would like to know what files I can be posting for a better understanding of the problem. I read the official wiki, it seems that one way audio is one of the main problems when using FreePBX, right? I followed all the steps described there and still the error continues)

When using VoIP. Not just FreePBX

1 Like

Ah yes. Sorry for the confusion.
@tm1000 what do you think I should post ?

Is there a firewall between the WebPhone and the PBX?

Hi, @PitzKey. Thanks for the reply.
No, there’s nothing. A call between the WebPhone(PJSIP) and a SoftPhone(SIP) works fine, but between Webphone don’t