The content in this article is appropriate for: Administrators
Latency, jitter, & packet loss can cause a decrease in quality for Voice Over IP (VoIP) phone calls. As such, it is vitally important to reduce these as much as reasonably possible.
The quality of a VoIP call is heavily dependent on the network environment. Factors include the device the client is running on, the network characteristics and firewall/router configuration and more. A VoIP deployment requires careful consideration of the end to end experience. This document is intended to share the best practices in configuring and selecting the best environment for VoIP calling.
Local network conditions have the biggest impact on voice quality. Packet loss, most frequently jitter-induced packet loss can cause the biggest impact. WiFi can be particularly bad for creating jitter.
Packet loss is very common in IP networks, but certain networks such as WiFi can be particularly prone to high levels of packet loss. This causes sections of media to be missing, and can cause the ‘robot’ distortion effect of media.
Callers start to notice the effect of latency around 250ms, above ~600ms the experience is unusable. There will always be some latency, the objective is to minimize it and keep total trip time well below 250ms.
Ideally latency should be below 100ms because while it's noticeable at 250ms, other services and issues beyond your control might add delay causing the cumulative total to be over 250ms.
Handling calls over an IP network requires the network to be in perfect working order. SingleComm has a recommended network and firewall configuration to minimize issues.
Please refer to this document for more information on how to configure for use with SingleComm: Call Center: Firewall and network configuration
Jitter is when packets don’t arrive in the same order they were sent.
Strategies to minimize jitter include:
Use fixed ethernet not WiFi wherever possible
Reduce packet conflicts on WiFi by reducing number of devices operating on the same channel.
Avoid large data file transfers going over the same WiFi environment concurrently with voice.
Avoid bufferbloat. This occurs when your router is unable to transmit all the packets required, and builds up too large a queue (rather than dropping packets when the queue length starts making latency noticeable). This queuing causes large latency and bursts of jitter. The real-time nature of voice means that this is not helpful.
Configure your router with a low buffer size.
For small amounts of jitter, this can be resolved in the jitter buffer – a queue of media packets waiting to be played which can be shuffled into the correct order while they wait in the queue. The length of the jitter buffer introduced must be traded off against the impact of increased latency. Too much jitter cannot be resolved by a reasonable length jitter buffer without introducing too much delay, so instead results in jitter induced packet loss causing choppy audio.
Latency and Delay are similar terms that refer to the amount of time it takes a bit to be transmitted from source to destination.
Latency is the time it takes the RTP (media) packets to traverse the network. Too much latency causes callers to speak over the top of each other.
Jitter is Delay that varies over time or when packets don’t arrive in the same order they were sent.
Supplemental & Related information:
Additionally, in order to check your overall firewall and port configuration, we recommend: UDP Port Scan with Nmap
Additional tools:
Check your connection quality with a tool such as these: