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Updated in 4.45
Call Recording Node

Description

The Call Recording Node starts, stops, pauses, or resumes recording of the current call. Using the SIPREC protocol (RFC 7866), Voice Gateway streams a copy of the call audio in real time to a session recording server (SRS) that you operate. The SRS can be any server that supports the SIPREC standard.
The Call Recording Node uses SIPREC only. For WebSocket-based recording, use the Call Recording setting in the Voice Gateway Self-Service Portal instead.

Parameters

SIP URI Formats

Supported formats

When multiple URIs are provided, Voice Gateway opens a parallel SIPREC dialog to each URI. The recording is considered successful if at least one dialog is accepted.
Voice Gateway accepts only SIP URIs. The following formats aren’t supported:
  • WebSocket URLs such as wss://… or ws://….
  • HTTP URLs such as https://… or http://….
  • Plain hostnames or IP addresses without a SIP scheme. For example, siprec.example.com or 10.0.0.5.

Troubleshooting

FAQ

No. A recorded call counts as one active session. Additional SIP dialogs and RTP streams are handled by the SRS. Those streams don’t count against Voice Gateway quotas, even when multiple SRS instances receive the audio.
Last modified on June 15, 2026