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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.

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Supported Protocols

The following protocols are supported:
ProtocolSupported Functionality
SIPSIP signaling, as specified in IETF RFC 3261.
TLSThe Transport Layer Security (TLS) is a cryptographic protocol that provides secure communication over a network. Voice Gateway supports TLS 1.2 and later versions. TLS 1.3 is preferred for optimal security.
SIP over TLSPartially encrypted SIP, as specified in IETF RFC 3261. Some servers may communicate unencrypted, depending on specific requirements.
SIPSFully encrypted SIP, as specified in IETF RFC 3261. All servers in the communication chain use TLS for encrypted communication.
SIP over WebSocketsSIP over secure WebSockets, as specified in RFC 7118.
RTPThe Real-time Transport Protocol as specified in RFC 3550.
SRTPEncrypted media using the Secure Real-time Transport Protocol (SRTP), as specified in RFC 3711.
RTCPThe 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.
DTMFThe use of RTP payloads to carry DMTF events, as specified in RFC 2833.
SIP Digest AuthenticationSIP Digest Authentication challenges and authenticates SIP devices, as specified in RFC 8760.
SIPRECActing as a SIPREC client or server to accomplish call recording, as specified in RFC 7866.
WebRTC clientsReceiving calls from WebRTC clients, such as web browsers or native mobile apps.
DTMF - SIP INFOReceiving DTMF via SIP INFO, as specified in RFC 2976.
SIP OPTIONSOPTIONS pings to allow remote SIP gateways and Session Border Controllers (SBCs) to test the health of the Voice Gateway SBCs.
Session timersSIP session timers, as specified in RFC 4028.
SIP UPDATEThe SIP UPDATE method to refresh SIP session.
STUNSession Traversal Utilities for NAT (STUN), as specified in RFC 5389.
Diversion Indication in SIPThe Diversion header, as specified in RFC 5806.
SIP REFERSending the SIP REFER method to transfer calls, as specified in RFC 3515. Receiving SIP REFER is not supported.

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Supported Codecs

The following audio codecs are supported:
  • G.711 (preferred):
    • A-law.
    • U-law.
  • OPUS.
  • G.722.

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Features

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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.

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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.
  • Configurable Atmosphere Sounds playing in the background of the conversation.
  • Configurable Silence Overlay playing background sounds when there’s a long pause in a call.

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Call Management

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STT and TTS providers

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Call Recording and Storage

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Call Quality Monitoring and Optimization

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Technical Support

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