How to Configure Network Requirements and Optimisations for Telviva Mobile and Telviva One

Definitions & Key Performance Indicators (KPIs)

• Network Mean Opinion Score (MOS): The metric evaluating the network's impact on the listening quality of a VoIP conversation, rated on a scale from 1 (poorest) to 5 (highest quality).

• Latency Benchmark: Callers begin noticing delays when latency exceeds 250 ms. Latency above ~600 ms makes calls virtually unusable.

• Packet Loss Rate: The percentage of transmitted packets that fail to reach their destination. High packet loss leads to audio gaps and cut-outs.

• Inter-arrival Jitter: The variation in packet arrival times measured in milliseconds (ms). Excessive jitter causes robotic voice distortion.

• UDP Timeout Benchmark: NAT UDP timeout must be configured to at least 600 seconds to preserve SIP session state.

Overview

This article explains how to configure network settings, firewall rules, and Wi-Fi parameters for Telviva Mobile and Telviva One. This process is used for optimizing local network conditions for voice traffic and helps you eliminate dropped calls, audio distortion, long pauses, and latency issues.

FAQ

What is Telviva Network optimisation?

Telviva Network optimisation is the process of setting up network firewalls, routing parameters, and Wi-Fi environments to prioritize voice traffic and ensure high Quality of Service (QoS) across Telviva platforms.

Why should I Optimize My Network for Telviva Services?

Local network issues like latency, jitter, and packet loss are the primary causes of degraded VoIP audio. Proper network configuration ensures clear voice quality, reduces dropped calls, and maintains reliable call signaling.

Network optimisation Process

Phase 1: Firewall and Router Configuration

Allow outbound and inbound VoIP traffic to traverse company firewalls and routers to Telviva hosted platform.

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Ports:

• Open outbound Port 4433 (TCP/UDP Outbound direction).

• Open outbound Port 3478 (TCP/UDP Outbound direction.

• Open outbound Port 5349 (TCP/UDP Outbound direction).

• Open Ports 32800 – 39999 (UDP bidirectional ).

• Disable SIP ALG on the firewall/router.

• Set the UDP timeout to at least 600 seconds to maintain consistent SIP traversal through NAT.

• Set DSCP marking to 46 (EF - Expedited Forwarding, Voice) for RTP media packets.

• Ensure device access is granted to [https://one.telviva.com/api](https://one.telviva.com/api) for API interactions.

Phase 2: Routing and Network Environment Tuning

• Ensure firewall NAT timers are set to 600 seconds for sustained UDP connections.

• Allow outgoing and incoming UDP to the public internet.

• Permit HTTPS, WebSocket, and Secure RTP traffic.

• Configure routers with a low buffer size to prevent buffer bloat, which induces latency and jitter bursts.

• Optionally configure router Quality of Service (QoS) rules to prioritize media traffic within the designated UDP port ranges.

Phase 3: Wi-Fi Environment Best Practices

• Perform a pre-installation RF survey to ensure 5 GHz coverage with at least -67 dB signal strength across all voice areas.

• Create a dedicated SSID specifically for VoIP devices.

• Set the SSID authentication type to 'Pre-shared key with WPA2'.

• Enable '5 GHz band only' on the dedicated voice SSID.

• Enable 'Traffic shaping' on the SSID to prioritize voice traffic.

• Set NAT UDP timeout to 600 seconds.

• Allow SIP port 5060 for both UDP and TCP traffic.

• Allow RTP ports 10000–20000 (UDP) on the local router.

• Allow UDP port 65550 if Vibe traffic passes through the firewall.

Phase 4: Vibe Implementation (Desk Phones Only)

Note: Vibe is not required for Telviva One Mobile or WebRTC deployments. It is only required for sites using desk phones.

• Deploy the dedicated Telviva router provided as part of the managed solution.

• Route Vibe traffic directly through the customer site router rather than customer firewalls, where corporate policy permits.

• Ensure routing to the Telviva hosted platform as per the IP blocks above.

• Set DSCP marking to 46 (EF - Expedited Forwarding, Voice) for RTP.

Troubleshooting

Problem: Callers experience delay and talk over each other (High Latency)

1. Solution:

• Upgrade low-bandwidth fixed internet connections to higher capacity links.

• Avoid using mobile networks (e.g., LTE/4G Data) for primary VoIP connectivity due to inherent high latency.

• Ensure latency remains below 250ms across network hops.

• Addendum: Latency is the time required for RTP (media) packets to travel to their destination.

Problem: Audio gaps, missing words, and cut-outs (Packet Loss)

2. Solution:

• Reduce Wi-Fi packet conflicts by redistributing devices across different Wi-Fi channels.

• Prevent large background data transfers on the same Wi-Fi network during active calls.

• Configure low router buffer sizes to prevent buffer bloat.

• Apply QoS rules prioritizing RTP traffic.

• Addendum: Packet loss represents data packets that fail to reach their destination during transmission.

Problem: Robotic distortion or choppy audio (Inter-arrival Jitter)

3. Solution:

• Isolate voice traffic to a dedicated 5 GHz SSID with active traffic shaping.

• Verify router UDP timeout is configured to 600 seconds.

• Ensure jitter buffers are not overwhelmed by extreme network packet arrival variations.

• Addendum: Jitter is the variation in packet arrival times caused by network congestion or wireless interference.