• Jacek Jarmulak

SIPREC support in Voicegain platform

Updated: Dec 22, 2020



Voicegain Speech-to-Text and Speech Analytics platform supports SIPREC protocol as one of the ways an audio stream of a telephone call can be fed to the speech recognizer.


The Session Recording Protocol (SIPREC) is an open SIP-based protocol for call recording. The standard is defined by Internet Engineering Task Force. It is supported by many phone platforms and call recording system vendors.


The SIPREC standard defines a protocol used to interact between a Session Recording Client (the role generally performed by PBX system or Session Border Controller) and a Session Recording Server (a third party call recorder, in our case a Voicegain-provided SIPREC server). SIPREC opens two RTP streams (one for inbound and one for outbound audio of the call) to the Recording Server. SIPREC protocol also is able to transfer call metadata to the Recorder, this is important so that the recordings can be tied to the information about the calls.


Use Cases


SIPREC is usually used for call recording but the standard essentially provides a real-time audio stream from the telephone call which makes it suitable for applications which have to work real-time like, e.g., agent assist or agent monitoring. Using the SIPREC interface Voicegain can provide real-time transcript of the call as well as perform speech analytics tasks in real time, e.g., keyword and phrase detection, personally-identifiable information scrubbing, sentiment and mood estimation, named-entity recognition, and variety of metrics (like silence, overtalk, etc.).


Audio obtained via SIPREC can also be recorded and transcribed, analyzed, or retrieved at a later time.


Supported Clients


Voicegain SIPREC interface has been tested with the following platforms:

Voicegain can capture relevant call metadata in addition to obtaining the audio (the metadata capture functionality may differ in capabilities depending on the client platform).


Voicegain platform can be configured to automatically launch transcription and speech-analytics as soon as the new SIPREC session gets established.


The output from transcription and speech analytics is available via a Web API. We also support websockets for more convenient streaming of the transcription and/or speech analytics data. SIPREC support is available both in the Cloud and the Edge (OnPrem) deployments of the Voicegain Platform.


SIPREC is an Enterprise feature of the Voicegain platform and is not included in the base package. Please contact support@voicegain.ai or submit a Zendesk ticket for more information about SIPREC and if you would like to use it with your existing Voicegain account.


Notes about Genesys Platform


Genesys Voice Platform does not support SIPREC directly. However, it does support streaming of the inbound and outbound RTP media to two separate SIP endpoints - the end result being pretty much the same as if SIPREC was used. We are currently working on implementing support for this feature of the Genesys Voice Platform for real-time audio streaming to Voicegain Platform. It should be available in Q1 2021.

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