Real Time Text Issues

I am having an issue when enabling Real-Time Text between two phones. These phones are custom phones that have the protocol T140 enabled. I have enabled T140 in the sip.conf file as well as the textsupport flag. When creating a call between the two phones, I can see the m values in a packet capture that shows all of the proper ports for audio and text and that the text protocol being used is T140. Once the call is placed, text starts flowing on both phones with repeated RTP (ex. RTPRTPRTPRTPRTPRTPRTP). This goes for the whole call.

I have noticed that if I try sending text, it will go through to the other phone and vice-versa. There is just a lot of gibberish with the RTP to really see it on the device. Looking at a packet capture, I can see the RTP text and my text going through. The RTP text looks like it only originates from the freepbx server whereas the text I send originates from the phone, like I’d expect.

Is the RTP message from freepbx some type of keep-alive that the phones are supposed to filter or am I missing something in my settings?

There was an issue[1] whereby the text was not null terminated. This is fixed in latest releases and may be the issue.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-28974

I have updated to the latest version but an still seeing the issue. I don’t know if that fix has been applied yet or not though. This issue is looks similar but it looks like this user is sending a message and then they see the RTP come across. What I am seeing is when the call is answered, the letters RTP start being sent to both devices continuously until the call is ended. When I type a message, I can see the letters pop in (ex. RTPRTPRTPTRTPeRTPsRTPt, the work Test in there). Looking at the packet capture, I see the T140 RTP start as both phones sending an empty packet (with the payload being UTF-8 or efbbbf). Then after a few of those, the FreePBX server just starts sending the letters RTP over and over. Just curious as to why it would be doing that.