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For an industrial environment with an average load of ap.100 thousand calls up to 400 thousand calls per day (250-350 simultaneous calls with call recording) and about 100 thousand chats per day, it is necessary to prepare 3 virtual machines, which are described in paragraphs 1-3 of the table below. You just need to install the operating system and leave Internet access to be able to download and install all additional software. After installation is complete, the Internet access from virtual machines can be disabled. Access to the servers is provided to developers via SSH with the root user permissions for the time which is needed for the applications installation and testing. Virtual machines 1-3 should be able to operate on the same VLAN without filtering traffic by ports (allow all).

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ObjectSoftwareResources (min)ConnectivityDescription
1. Webitel Server Telephony application server

Debian 10 12 Linux 64bit - — basic server installation with SSH connection;

Additionally, we will install:

OpenSIPS 3.4; rtpengine; nginx; SSL; webitel 

  • 4 vCPU > 2,4 6 GHz (Haswell generation or newer), No more than 2 virtual cores per 1 physical core
  • 8Gb RAM
  • 120 60 GB, 20 1000 IOPS, <10 мс 
  • 1 network interface >= 1 GbE

in-out: 80/tcp, 443/tcp, 5060/tcp, 5060/udp

in-out: 10000-50000/udp

Telephony server. SSL certificates with a trusted CA are required to provide a secure connection via HTTPS protocol and telephony in a browser..


2. Database / Cache Server / Storage Server

Debian 10 12 Linux 64bit - — basic server installation with SSH connection;

Additionally, we will install:

Linux; PostgreSQL 1215; Grafana; Consul; RabbitMQ; Webitel Storage 
  • 8 vCPU, > 2,6 8 GHz (Haswell generation or newer), No more than 2 virtual cores per 1 physical core
  • 16Gb RAM
  • 250+ 60 GB, 100 1000 IOPS, <8 мс1 TB, 10 IOPS, <10 мс (зависит от количества, продолжительности и времени хранения записей разговоров)
  • 250 GB, 2000 IOPS
  • 1 network interface >= 1 GbE



  • Database.
Call recording storage
  • Microservice registration service.
  • Messaging bus.

We recommend duplicating the server to configure Streaming Replica.

3. Media Gateway Application Server

Debian 10 12 Linux 64bit - — basic server installation with SSH connection;

Additionally, we will install:

FreeSWITCH 10; Webitel FlowManager; Webitel App; Webitel API GW; Storage; CallCenter; Engine; Messages
  • 8 vCPU, > 2,6 GHz (Haswell generation or newer), No more than 2 virtual cores per 1 physical core
  • 16Gb RAM
  • 120 GB, 20 80 GB, 1000 IOPS
  • 1 TB, 200 IOPS, <8 мс<10 мс (if S3 is not used)
  • 1 network interface >= 1 GbE

Voice menu server.


  • Applications server.
  • Server for creating voice menus.
  • Storage of conversation recordings.
4. External Data Services


Взаимодействие с внешними системами происходит по средствам HTTP REST (нужно обсудить)Interaction with external systems runs via HTTP REST.
5.  Team

Microsoft Windows 10 11 / Linux 64bit

Google Chrome / Microsoft Edge (the latest or previous version)

  • 4 vCPU
  • 8Gb RAM
Connection to the IS not less than 2 Mb/s per user, with delays of no more than 15-20 ms.Employees’ workplace
6. Hardware SIP Phone

Connection to the IS not less than 5 Mb/s per device, with delays of no more than 15-20 ms.

in-out: 5060/udp, 10000-20000/udp

Hardware phone with SIP 2.0 protocol support
7. PSTN


Public Switched Telephone Network is a general subscriber communication network, the access to which is provided by telephone sets, PBX, and data transmission equipment.



Connection speedDelay (Ping)JitterPacket Loss

WebRTC

  • For audio calls: at least 100–300 Kbps.
  • For video calls (standard quality): at least 1–2 Mbps.
  • For high-quality video (HD or Full HD): at least 2–4 Mbps.
  • For conferences or streams (4K video): 8–25 Mbps is recommended.
  • Optimal delay for WebRTC should be less than 50 ms.
  • Acceptable delay: 50-100 ms.
  • If delay exceeds 150 ms, noticeable delays during calls or video may occur.
  • Optimal jitter should be less than 30 ms. Jitter refers to variations in packet delay, and high jitter can significantly degrade audio or video quality.

  • Jitter over 50 ms can cause distortion or delays in audio/video.


  • Optimal packet loss for WebRTC: less than 1%.
  • Loss of up to 2-3% may be acceptable for audio, but it will degrade video quality.
  • Loss above 5% will significantly impair communication, causing noticeable interruptions, artifacts, or video 'freezes'.