CPaaS API Exposure Function

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.

Features & Benefits

Super lightweight all-in-one cloud native solution

Enables fast and easy automated deployment, instant configuration, zero-day service and secured isolated environment for each tenant

High-density media processing on cloud environment

Supports up to 1,000 channels of G.711 RTP with voice play or record on containerized instances

High-density media processing on cloud environment

Supports up to 1,000 channels of G.711 RTP with voice play or record on containerized instances

Enhanced  built-in security tools

Provides encryption protection at the media layer, signaling layer and control layer. Built-in firewall for calling and messaging

Advanced multimedia feature set

Voice features, such as wideband audio codec support, play, record, transaction record, DTMF detection, and Call Progress Analysis

CPaaS Cloud Communication Engine

Rich set of communication APIs

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

Calls

Make and receive voice calls and perform media tasks on calls.

Messages

Send and receive instant and short text messages to groups or individuals.

Bridges

Create bridges with several voice calls and perform basic media tasks on bridges.

Administration

Get/Set system configuration, manage file system and control service state.

Contacts

Create and manage list of contacts and groups, monitor presence and state.

Standard-based Connectiviy Options

SIP trunks

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.

A Multitude of Use Cases

Traditional phone systems

Auto attendant

Use an automated voice menu system that allows callers to be transferred to a destination without going through a telephone operator or receptionist. 

The voice menu can also be used to collect information on the caller or to a first classification of the caller to efficiently route the call to a matching destination.

Interactive voice response

Use automated voice menus to collect information on callers or callee. Record audio clips and build call flows guided by user input via the phone keypad.

Commonly used for applications live voicemail, license activations, credit card charging and more...

Call recording

Network-based call recording that doesn't required the smartphone with an application to record a call between 2 parties.

Oftenly use in call centers applications for customer satisfaction and quality of service rating.

Private branch exchanges

Remote extensions

Enable seamless remote working and smooth call transition from the office branch to browser-based externsions.

Remove the complexity of the PABX remote extensions connectivity and security by enabling the remote extension using the WebRTC protocol.

Presence

Enable your application with the ability to monitor office branch extensions statuses and presence.

Commonly use, for example, in a contact center environment to monitor the agents availability.

Group messaging

Implement instant messaging in your working environment for seamless team communication.

Share text messages, multimedia files and screen clips without transiting to a 3rd party application.

In-app web and mobile

Click to call

Make a direct call from your web page or mobile application to your sales or support department to engage with customers.

WebRTC bridge

Make direct VoIP calls from your web or mobile apps to telephone destinations without using the handset stock dialer.

Connect any combination of pstn calls, WebRTC calls or internal PABX calls together in one bridge.

Commonly use for 3-way calls, call recording or conference bridges.

Two-factor authentication

Add an extra layer of security to your web or mobile application to make sure that people trying to gain access are who they say they are. 

Technical specifications

SIP stack

RFC 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

Multimedia codecs

G.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

RTP stack

RFC 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

WebRTC stack

RFC 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

HTTP stack

RFC 2616 | HTTP/1.1: Hypertext Transfer Protocol

RFC 7230 | HTTP/1.1: Message Syntax and Routing

RFC 7231 | HTTP/1.1: Semantics and Content

RFC 7232 | HTTP/1.1: Conditional Requests

RFC 7233 | HTTP/1.1: Range Requests

RFC 7234 | HTTP/1.1: Caching

RFC 7235 | HTTP/1.1: Authentication

RFC 5849 | OAuth 1.0: Authorization Protocol

RFC 6749 | OAuth 2.0: Authorization Framework

RFC 6750 | OAuth 2.0: Bearer Token Usage