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Voice Gateway Technical Capabilities

This article is intended for individuals considering Voice Gateway for their telecommunication needs. It provides a detailed overview of the platform's extensive technical capabilities, including support for various protocols such as SIP and RTP, as well as audio codecs. The description highlights several features, including call routing, custom SIP headers, and call recording integration, which address diverse requirements. Additionally, the article describes the use of STT and TTS providers and emphasizes the advanced tools for monitoring call quality.

Supported Protocols

The following protocols are supported:

Protocol Supported Functionality
SIP SIP signaling, as specified in IETF RFC 3261.
SIPS SIP over TLS signaling, as specified in IETF RFC 3261.
SIP over WebSockets SIP over secure WebSockets, as specified in RFC 7118.
RTP The Real-time Transport Protocol as specified in RFC 3550.
SRTP Encrypted media using the Secure Real-time Transport Protocol (SRTP), as specified in RFC 3711.
RTCP The RTP Control Protocol (RTCP) provides quality of service (QoS) statistics for RTP media streams.
Session Description Protocol (SDP) The Session Description Protocol (SDP), as specified in RFC 4566.
Offer/Answer Model with the Session Description Protocol (SDP) The offer or answer model for negotiating SIP sessions, as specified in RFC 3264.
DTMF The use of RTP payloads to carry DMTF events, as specified in RFC 2833.
SIP Digest Authentication SIP Digest Authentication challenges and authenticates SIP devices, as specified in RFC 8760.
SIPREC Acting as a SIPREC client or server to accomplish call recording, as specified in RFC 7866.
WebRTC clients Receiving calls from WebRTC clients, such as web browsers or native mobile apps.
DTMF - SIP INFO Receiving DTMF via SIP INFO, as specified in RFC 2976.
SIP OPTIONS OPTIONS pings to allow remote SIP gateways and Session Border Controllers (SBCs) to test the health of the Voice Gateway SBCs.
Session timers SIP session timers, as specified in RFC 4028.
SIP UPDATE The SIP UPDATE method to refresh SIP session.
STUN Session Traversal Utilities for NAT (STUN), as specified in RFC 5389.
Diversion Indication in SIP The Diversion header, as specified in RFC 5806.

Supported Codecs

The following audio codecs are supported:

  • G.711 (preferred):
    • A-law.
    • U-law.
  • OPUS.
  • G.722.

Features

Trunk management and Routing

  • Transferring calls via SIP REFER or SIP INVITE.
  • Multiple SIP trunks per customer.
  • Configuring SIP trunks with options like tech prefix, SIP Diversion header, Outbound authentication (including REGISTER).
  • Routing calls based on a trunk group, Direct Inward Dialing (DID), or DID range.
  • Least-cost routing selection of outbound trunk.

Call Features

  • Custom SIP headers on inbound and outbound calls.
  • Mid-call SIP INFO requests.
  • P-Asserted-Identity header to identify caller.
  • Receiving compact SIP headers.
  • Receiving re-INVITE with no Session Description Protocol (SDP).
  • Answering machine detection.
  • Configurable music on hold.

Call Management

STT and TTS providers

Call Recording and Storage

  • Call recording via integration with a third-party SIPREC server.
  • Pausing or resuming transcription and recording to avoid the collection of Personally Identifiable Information (PII).
  • Call recording with no third-party components, recordings streamed to the preferred cloud storage.

Call Quality Monitoring and Optimization

Technical Support

For any further or more specific questions about the Voice Gateway technical capabilities, contact Cognigy technical support.