EN 301 549 6.2 -- Real-Time Text (RTT)
What It Is
EN 301 549 v3.2.1 clause 6.2 says that where ICT provides two-way voice communication, it must also provide a means for two-way real-time text (RTT) communication on the same connection, and voice and RTT must be usable concurrently[1]. RTT transmits each character the moment it is entered into the send buffer, so the receiver watches the message form as the sender types -- the text analogue of how voice transmits sound continuously, not in chunks the way chat or SMS does.
The clause pins down the protocol stack by reference. For SIP/VoIP endpoints, the applicable mechanism is IETF RFC 4103, which defines an RTP payload format for text conversation based on ITU-T Recommendation T.140[2]. T.140 itself specifies the character presentation layer -- the ISO 10646 code set, the control codes, and the rules for how characters flow into the receiver's display[3]. For PSTN-connected ICT the clause points at ITU-T V.18 instead. Clause 6.2.4 also sets a transmission latency requirement: RTT input has to reach the network within 500 ms of being available to the transmitter, excluding platform and network delay[1]. Together, voice + RTT + video is what ETSI and ITU-T call "Total Conversation," and 6.2 is the text leg of that stool.
Why It Matters
Deaf, hard-of-hearing, deafblind, and speech-disabled users use RTT in place of -- or alongside -- voice on the same call, with the same hearing party on the other end. Because characters appear as they are typed rather than after a send, both sides can interrupt, overlap, and read pauses the way voice callers read pauses. A chat pane bolted onto a voice client does not achieve this: chat is turn-based and the receiver sees nothing until Enter is pressed, which collapses the conversational rhythm that RTT preserves. RTT is also the IP-era replacement for TTY/textphone hardware, which is being decommissioned as carriers retire TDM voice paths.
How It Relates to WCAG
WCAG has nothing to say about voice calls -- its scope is web content, not telephony protocols. Clause 6.2 is one of the layers in EN 301 549 that goes past WCAG: chapter 9 of EN 301 549 adopts WCAG 2.1 Level AA for web content, but chapters 6 through 8 cover ICT that WCAG never touches[1]. Video-meeting and VoIP vendors entering the EU routinely pass the chapter 9 web requirements and then fail 6.2 at the protocol layer, because the failure is not in the UI but in the media stack -- there is no T.140 stream in the SIP offer and no RFC 4103 payload on the wire.
Practical Implications
- Soft-phones and VoIP clients must negotiate an RFC 4103 text stream in SIP, not expose a separate chat widget as a substitute.
- RTT and voice must be usable simultaneously on a single call, not as alternative modes.
- Character transmission latency must stay inside the 500 ms budget set by clause 6.2.4 (excluding network delay).
- Emergency-call (112) handling has to carry RTT end-to-end, so the PSAP side must also accept the stream.
- Send and receive text must be visually and programmatically distinguishable, per clause 6.2.2.
- The RTT interface itself also has to meet chapter 9 / WCAG for contrast, text sizing, and keyboard operation -- adding the protocol is necessary but not sufficient.
The Fix
6.2 is a protocol-stack addition, not a UI change. For SIP-based clients that means adding a text media line to the SDP offer, implementing the RFC 4103 RTP payload format (including the T.140 redundancy scheme RFC 4103 defines for packet-loss tolerance), and wiring a T.140 code path into the client's media engine so the received character stream drives a live display. Expect the work to cross the media stack, the signalling layer, and the call UI -- it is not a ticket a front-end team can close alone. Certification against the clause means interop testing with at least one other RFC 4103 endpoint, because an isolated implementation that only talks to itself does not satisfy the "two-way" requirement.
Related Clauses
Sources
- ETSI EN 301 549 v3.2.1, clause 6.2
- ITU-T Recommendation T.140 (1998)
- IETF RFC 4103 -- RTP Payload for Text Conversation (2005)