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Cloud Native

CPaaS

Communication Platform

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built by slicce

The all-in-one enterprise communications microservice

The CPaaS API Exposure Function is a communication engine that enables enable customers to develop enterprise communications services in a more secure, more customizable and more cost effective way.

  • Superlight cloud native solution
  • Built-in security features
  • Advanced multimedia feature set
  • Media processing on cloud resources

Rich business communication API set

Programmable Voice and Programmable SMS for public & private branch exchanges.

  • Calls API: Make and receive voice calls and perform media tasks on calls.
  • Messages API: Send and receive instant and short text messages to groups or individuals.
  • Bridges API: Create bridges with several voice calls and perform basic media tasks on bridges.
  • Contacts API: Create and manage list of contacts and groups, monitor presence and state.
  • Admin API: Get/Set system configuration, manage file system and control service state.

Standard-based Connectiviy Options:

  • SIP trunking: Connect one or more SIP trunks to the engine to gain the ability to service inbound and outbound VoIP calls.
  • SIP extensions: Connect the engine to a class 5 soft-switch or to a private branch exchange to gain the instant messaging and presence capabilities in addition to the voice calls.
  • SMPP: Connect to an SMPP server, to gain the ability of sending short text messages to mobile destinations.
  • WebRTC: Use the WebRTC interface to accept or make calls from web-browsers using the in-browser stack delivered in Chrome, Edge and Firefox.
Cloud Native api exposure functions CAMEL gateway DIAMETER gateway SIP gateway MAP gateway USSD gateway SMS gateway SMS and Charging Interworking functions cloud native SS7 load balancers cloud native diameter load balancer cloud native sip load balancer cloud native GTP load balancer

Per customer instances brings secured private instant CPaaS capabilities to customers

Slicce CPaaS microservice offer businesses a flexible, scalable, and cost-effective standalone solution for their communication needs. The service simplifies deployment, offers rich customization options, and provides businesses with the tools to deliver reliable, secure, and personalized communication experiences to their customers. With global reach, high availability, and integrated analytics, Slicce CPaaS enables businesses to stay competitive and deliver enhanced customer experiences while focusing on their core operations.

Fully managed cloud native service Fully managed cloud native service

functional specs

SIP stackRFC 3261 | SIP: Session Initiation Protocol RFC 3262 | SIP Reliability (PRACK) RFC 3263 | SIP: Locating SIP Servers RFC 3264 | SDP Offer/Answer RFC 3265 | SIP Specific Event Notification RFC 1321 | MD5: Message Digest Algorithm RFC 2617 | HTTP Authentication RFC 2806 | URLs for Telephone Calls RFC 2833 | RTP Payload for DTMF & Tones RFC 2915 | NAPTR: Naming Authority Pointer RFC 2976 | SIP INFO Method RFC 3204 | MIME Objects for ISUP and QSIG RFC 3310 | HTTP Digest Authentication – AKA RFC 3311 | SIP Update Method RFC 3329 | Security Mechanism for SIP RFC 3428 | SIP Extension for IM RFC 3489 | STUN: Simple Traversal UDP - NATs RFC 3515 | SIP Refer Method RFC 3581 | Symmetric Response Routing Ext’n RFC 3665 | SIP Basic Call Flow Examples RFC 3711 | SRTP Secure RTP RFC 3891 | SIP ‘ Replaces’ Header RFC 3903 | SIMPLE SIP for IM and Presence RFC 4028 | Session Timers in SIP RFC 4346 | TLS Transport Layer Security RFC 4566 | SDP Session Descrip’n Protocol/IPv6 RFC 4568 | SDP Security for Media Streams
RTP stackRFC 1889 | RTP: A Transport Protocol for Real-Time Applications RFC 1890 | RTP Profile for Audio and Video Conferences with Minimal Control RFC 2508 | Compressing IP/UDP/RTP Headers for Low-Speed Serial Links RFC 3550 | RTP: Real-Time Transport Protocol RFC 3551 | RTP Profile for Audio and Video Conferences with Minimal Control RFC 3711 | SRTP: The Secure Real-time Transport Protocol RFC 2833 | RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
Multimedia codecsG.722 | Standard 7 kHz wideband audio codec G.729 | Standard compressed narrow-band VoIP audio codec G.711 | Standard non-compressed narrow-band audio codec VP8 | Google video compressed video codec
WebRTC stackRFC 7478 | Web Real-Time Communication Use Cases and Requirements RFC 7675 | Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness RFC 7742 | WebRTC Video Processing and Codec Requirements RFC 7874 | WebRTC Audio Codec and Processing Requirements RFC 7875 | Additional WebRTC Audio Codecs for Interoperability

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