Integrating TTY over VOIP

Hi guys I am trying to integrate TTY over VoiP. Do I have to implement special measures in-order to support TTY over VoiP or the configuration would be on the part of the SIP phone? Any information would be helpful. Thank you.

This seems to be basically the same question as TTY over VOIP - General Help - FreePBX Community Forums and again I’d suggest that TDD is less confusing than TTY, because the latter term was used for mechanical teletypes (and is the origin of tty in the names of Unix/Linux serial ports…

A standard SIP phone would not know what to do with TDD modem tones, so your first thing would be to decide on what phones you are going to use. I imagine that most are analogue phones, so you will need an ATA. As said in the FreePBX thread, the complete path to the PSTN needs to be G.711 (3.1kHz audio, not speech) all the way to the PSTN.

I do not believe that Asterisk supports RFC 4103 media types, so communication would need to be based on analogue modem standards. Any system using 4103 would require a gateway to the analogue modem standards, in order to use the PSTN.

Wikipedia reports the existence of at least one SIP softphone with TDD support, both via modem tones and RFC 4103. IpTTY - Wikipedia

I’m basically researching this on the fly, so you should have access to the same sources.

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