Overview
The connection is via HTTPS with a level 3 certificate via (https://xxxx.ucontactcloud.com), uContact uses HTTPS (http2) with a REST API and Secure Websockets (wss).
The connection is made through Secure Websockets https://xxxx.ucontactcloud.com:8089/ws (for SIP signaling).
Our STUN Server (stun.ucontactcloud.com)
Agent and Telephony server negotiate with STUN Server gathering ICE candidates.
After that the connection is made from the agent to the telephony server, bridging the communication with the telephony provider (via SIP, SIP TLS, SIP WebRTC).
The telephony server does the bridging to host all the recordings, transcoding, etc.
Codecs: We use alaw, ulaw, opus, vp8, h264. Al the transcoding part takes action in the telephony server.
QoS: We can do TOS and COS tag (on the server-side) the packets with multiple parameters in different actions (video, audio, text)
Security
Google Cloud Security Plus is our security layer with Force Brut attack detection with all communications encrypted.
The server prefers cipher suites supporting Perfect-Forward-Secrecy.
The server provides HTTP Strict Transport Security.
The server provides HTTP Public Key Pinning.
X-XSS-Protection
Hardware Sizing (On-Premise)
Size | Agents | Telephony | Backend | Database |
---|---|---|---|---|
Small | < 30 | 1 | 0 | 0 |
Medium | < 500 | 1 | 1 | 0 |
Big | > 500 | 1 | 1 | 1 |
Accepted Telecommunications Equipment
Digium or Sangoma Cards, Xorcom, Dinstar, KHOMP, SIP Gateways, SIP Hardphones.
The servers should be certified in Ubuntu, check: http://www.ubuntu.com/certification/server/
Communication
Component Versions
BBDD: Mysql: 8 or 5.7
WebServer: NGINX 1.17. 0 (latest)
OS: Ubuntu 20.04, 18.04, 16.04
Telephony: Asterisk Modificado branch 13
Frontend: HTML5 y JS
Backend: Java 8
Structure
/etc/IntegraServer/web/ IntegraPortalWS.war IntegraChannels.war IntegraGamification.war /forms/ /images/ /uContact/ /workflowdesigner/ /formsdesigner/ /tmp/ /webchatclient/ /etc/IntegraServer/reports/ /etc/IntegraServer/server/ IntegraServer.jar /jasper/*.jar /jdbc/*.jar /mail/*jar /others/*.jar /ssh/*.jar /ws/*.jar
Network requirements for VoIP
Bandwidth
With G.711 codec: 100 kbps per call
Latency for toll-quality
<100 ms total
Jitter
< 20 ms jitter
Packet loss
< 1 % for voice calls
Required recordings Storage (audio y video)
Duration | Required Storage for audio (.gsm) | Required Storage for video (.webm) |
---|---|---|
1 Minute | 100 kb | 1.5 MB |
10 Minutes | 1 MB | 20 MB |
20 Minutes | 2.8 MB | 44 MB |
30 Minutes | 3.8 MB | 100 MB |