Last updated: April 7, 2026

PSTN stands for Public Switched Telephone Network.

The PSTN is the traditional circuit-switched telephone network that has been in operation globally for over a century. It encompasses landline phones, the copper wiring infrastructure, and the switching centers that connect calls.

PSTN and WebRTC

WebRTC applications frequently need to interoperate with the PSTN to enable features like:

  • Dial-in to video conferences via phone numbers
  • Click-to-call from web pages to phone numbers
  • VoIP to PSTN bridging

To connect WebRTC to the PSTN, a gateway or SBC (Session Border Controller) is required. This gateway translates between WebRTC’s protocols (SRTP, DTLS) and the PSTN’s protocols (SIP trunking, G.711 audio), handling transcoding when necessary.

The PSTN uses narrowband audio (8kHz sampling rate), which is significantly lower quality than the wideband or fullband audio that WebRTC’s Opus codec supports. This means audio quality is always reduced when bridging to the PSTN.

In most cases, to bridge with PSTN, WebRTC will first be connected via SIP, which is used today as a programmable bridge towards PSTN.

Additional reading

  • CPaaS vendors commonly provide PSTN connectivity as part of their platform
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About WebRTC Glossary

The WebRTC Glossary is an ongoing project where users can learn more about WebRTC related terms. It is maintained by Tsahi Levent-Levi of BlogGeek.me.