Brief Description - Asterisk supports RTT (real time text) with an addition of a couple statements in the configuration files, at least using chan_sip. I had this working in chan_sip and using TIPCon1 soft-phone (TIPcon1 download | SourceForge.net). This call scenario was executed successfully as audio / text, video text and audio/video/text. Text meaning real time text as in ITU T.140. Another essential configuration on the soft-phone to make this work was to disable 401 unauthorized on the INVITE message, otherwise the TIPCon1 / SIPCon1 would not respond to an 401 unauthorized message!
This was possible by not providing any “secret” in the FreePBX extensions configurations.
However, using chan_PJSIP and following the same (blank secret) it does not disable the 401 unauthorized message, at least according to what I have experienced.
Question: Hoe do I disable the 401 unauthorized message in PJSIP? The reason being is the TIPCon1 soft-phone and other endpoints that would not respond to such a message.
Any input would be welcome!
NG911 and similar initiatives in Europe (NG112) have a real text requirement!
Would it be prudent to ask for PJSIP support these essential services for saving lives?
Also, there seems to be a fundamental difference in the way chan_sip and chan_pjsip treat an empty “secret”. Nevermind the FreePBX (we will direct inquiries accordingly but what if we keep the password empty in Asterisk files? Are we disabling the sending of 401 Unauthorized on outgoing local extensions? Not really, unless my extensions.conf is not doing it! So besides the support issue we need to disable the 401 Unauthorized (on outgoing INVITE) at will!
I don’t have any timeframe on supporting such a thing for Asterisk. A patch was contributed, and code review was done, but the reporter never followed up to take care of any of the feedback and noone picked up the review otherwise.
In PJSIP if auth is not required then an auth section should not be created. I don’t know what setting the password to empty would do, because that’s not how you’re supposed to disable it.
So a patch was contributed but is not really available to be officially deployed, I guess.
Depending on the direction. maybe I could contribute along these lines. I am currently employed in the NG911 business and recently developed an asterisk implementation (rhel8) to be used as a call generation tool. I had a difficult time in getting things work or not in PJSIP, however!