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The following are the prerequisites for setting up Voice recording solution under different deployment profiles. 

Type

Source Host

Port

Destination

Notes

HTTPS

voice-recording-solution

444

any

To access VRS front-end4 8 500 10FusionPBX

Voice Recording Solution

Hardware Sizing

For up to 100 concurrent agents

  • For Operating System and services: 8 vCPU,

  • 16 GB RAM,

  • 200 GB HDD

  • For recording files: Disk Space (in bytes) = [Number of Calls × average duration (seconds)× bit rate /8] x 2
    for example;
    Calls in a month: 30000
    Average duration: 3 minutes = 180 seconds
    Bit rate for G711: 64 kbps
    So space required for one month of calls = (30000 x 180 x 64000/8) x 2 = 864000000000 bytes = ~ 80.46 GBs

Software Requirements

  • Host Operating System - Debian

  • EFswitch (Only for EFCX)

  • Postgresql (Only for EFCX)

System Access Requirements

  • Administrative privileges (root) on the host machine are required to proceed with installation.

Port Utilisation

The local security policy and firewall should allow open communication on the following ports.

Bandwidth Requirements:

Audio:

  • Minimum Bandwidth: Approximately 150 kbps per stream.

  • Optimal Bandwidth: 150-200 kbps per stream to ensure high-quality audio without compression artifacts.

QoS Parameters for Both Client and Server:

  • Packet Loss:
    < 0.5%: Ensures high-quality audio without significant degradation. Packet loss beyond this threshold can lead to noticeable quality issues.

  • Jitter:
    < 30 ms: Keeps variations in packet arrival times low, reducing audio distortions. Maintaining jitter within this range is crucial for consistent quality.

  • Latency:
    < 100 ms: One-way latency within this range helps maintain natural conversation flow without noticeable delays. Lower latency is critical for real-time communication.

  • Traffic Prioritization:
    Audio data should be prioritized over other network traffic: Ensures that voice quality remains consistent, even under network congestion. Implementing QoS policies that prioritize audio packets can significantly improve call reliability and clarity.


Info

(lightbulb) Time Synchronisation

Communication between Voice Recording solution, Cisco CUCM, Cisco IP Phones (BIB enabled) carry timestamps. If the system dates and time are not synchronised the system can produce unpredictable results. Therefore, please make every effort to adhere to the following time synchronisation guidelines:

Voice Recording solution, Cisco CUCM, Cisco IP Phones (BIB enabled) should have their Time Zone and time configured properly according to the geographic region and synchronised. To configure the time zone, please see the instructions from the hardware or software manufacturer of NTP server. Voice Recording solution and Cisco CUCM should be synchronised to the second. This synchronisation should be maintained continuously and validated on a regular basis. For security reasons, Network Time Protocol (NTP) V 4.1+ is recommended.

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