Recording Service - Service Specifications

The iotcomms.io Recording Service provides a robust, scalable, and secure service for integrating call recording capabilities into telephony systems. Designed with flexibility and reliability in mind, the service adheres to key industry standards and offers broad compatibility with SIPRec-enabled ecosystems.

1. Supported SIP RFCs

   The Recording Service complies with the following SIP and media standards to ensure interoperability and reliability:

   • RFC 3261 - SIP: Session Initiation Protocol

     Defines the core protocol for initiating, maintaining, and terminating SIP communication sessions.

   • RFC 3263 - SIP: Locating SIP Servers

     Describes how SIP clients use DNS to locate SIP servers, ensuring redundancy and reliability.

   • RFC 3550 - RTP: Real-Time Transport Protocol

     Defines the standard for transmitting audio and video over IP networks in real time.

   • RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control

     Specifies the profiles and payload formats for RTP audio and video streams.

   • RFC 3711 - SRTP: Secure Real-Time Transport Protocol

     Establishes encryption and integrity protection for RTP media streams.

   • RFC 4566 - SDP: Session Description Protocol

     Specifies the format for multimedia communication session descriptions used in SIPRec.

   • RFC 7866 - SIPRec: Session Recording Protocol

     Defines the standard for SIP-based call recording, enabling seamless integration with telephony systems.

2. Supported Codecs

   The Recording Service supports a range of audio codecs to ensure compatibility and high-quality recording:

   • G.711 A-law

     A standard narrowband codec commonly used in telephony systems.

   • G.711 µ-law

     A widely adopted codec in North America for VoIP.

   • G.722

     A wideband codec offering better audio quality for recordings.

   • G.729

     A low-bandwidth codec suitable for constrained network environments.

   • OPUS

     A versatile codec supporting narrowband to fullband audio, ideal for high-quality applications.

3. Transport Protocols

   To ensure secure and reliable communication, the Recording Service supports the following transport protocols:

   • SIP over TLS

     Secures SIP signaling for robust communication.

   • RTP/UDP and SRTP/UDP

     Handles media transport with or without encryption, ensuring compatibility with different telephony systems.

4. Security Features

   The Recording Service incorporates advanced security measures to safeguard call data:

   • TLS Encryption for SIP Signaling

     Protects communication between the telephony system and the Session Recording Server.

   • SRTP for Media Streams

     Encrypts RTP media streams to ensure the privacy of recorded audio.

   • Encrypted Storage

     Supports encryption at rest using customer-specific keys, ensuring secure multi-tenant operation.

   • Secure Authentication

     Uses API keys and JWT tokens to authorize API interactions securely.

5. Integration Capabilities

   The Recording Service offers seamless integration with telephony systems and applications:

   • SIPRec Compatibility

     Tested with major SBCs like Metaswitch's Perimeta and Cisco's CUBE for broad adoption.

   • Amazon S3 Integration

     Natively supports storing recordings in customer-designated S3 buckets or S3-compatible on-premise storage.

   • REST API Control

     Enables advanced recording management, including starting, stopping, trimming, and masking sensitive data.

   • AWS SNS/SQS Events and Commands for Recording Management

     Provides real-time notifications and advanced command execution for lifecycle management and error handling.