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
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