To ensure the best experience for both your customers and guest who join the call, as well for the users who initiate a call, there are a few requirements we recommend fulfilling:

Network

Local network conditions have the most significant impact on voice quality. Jitter, latency, and packet loss can be the biggest contributors to voice quality issues in any VoIP network.

Latency

The time it takes the RTP (media) packets to arrive at the destination

Causes media delivery delays, callers may speak over the top of each other.

Packet loss

Packets that don’t make it to the final destination

Causes gaps and cut-outs in media, callers may not hear the other side.

Jitter

Packets that arrive at the destination out of order

Cause a ‘robotic’ distortion effect in media, or packet loss when overrunning the jitter buffer

Latency: High latency can substantially degrade a caller’s experience. While there will always be some latency between the codec algorithm, the jitter buffer, and network traversal, the goal is to keep this to a minimum. Here are some strategies to minimize latency on your network:

  • Some lower bandwidth fixed internet connections can often have higher latency. If possible, upgrade your internet connectivity.

  • Stick to high-bandwidth connections. Mobile networks such as LTE (mobile 4G Data) can often have high latency.

  • Ensure you’re using the most recent SnapCall version to take advantage of SnapCall Global Low Latency routing infrastructure.

Jitter: Packet loss, most frequently jitter-induced packet loss, can have a significant impact on your VoIP call quality. Wi-Fi can be particularly bad for creating jitter. Here are some strategies to minimize jitter on your network:

  • Use fixed ethernet instead of Wi-Fi whenever possible

  • Reduce packet conflicts on Wi-Fi by reducing the number of devices operating on the same channel.

  • Avoid large data file transfers over the same Wi-Fi environment concurrently with voice.

  • Avoid buffer bloat, which can result in high latency, and bursts of jitter. We recommend ensuring your router is configured with small buffer size, as a buffer cannot mask high jitter without introducing artificial delay, and often choppy audio.

Note: Not all routers allow for configuring buffer sizes, but some routers ship with defaults, which are not optimized for real-time VoIP networks. Open-source routers, enterprise-grade routers, and gamer-oriented routers are good candidates for providing the right configuration options and defaults.

If you have addressed the above issues and continue to have jitter related impact on your voice quality, you may consider configuring your router with QoS rules to prioritize Snapcall traffic on the above media UDP ports.

Bandwidth

When it comes to bandwidth, we recommend the following set up to guarantee a good experience to both your users and guests:

  • Latency should be less than 50 ms

  • Outbound signals from a participant in all situations must meet a 3.2 mbps bandwidth requirement.

  • Inbound signals depend on the number of participants:

    • 2.6 mbps with 2 participants

    • 3.2 mbps with 5 participants

    • 4.0 mbps with 10 participants

These are the minimum requirements that guarantee a normal functioning of Snapcall within a browser:

  • Latency should be less than 100 ms

  • Outbound signals from a participant in all situations must meet a 1 mbps bandwidth requirement.

  • Inbound signals depend on the number of participants:

    • 1 mbps with 2 participants

    • 1.5 mbps with 5 participants

    • 2 mbps with 10 participants

Firewall configuration

You can find IP port and bandwidth requirements here.

Browser

Snapcall works best with Google Chrome, Apple Safari, Mozilla Firefox and Microsoft Edge. Other browser might be fully or partially compatible but not recommended for a daily usage.

When you start using Snapcall, please ensure you have installed the latest and up to date version of your browser. This can be checked directly on the settings of each browser.

We test with the current major revision of the browser, and the previous major browser revision. That is, if the current release version is N, we test with N and N-1. We also proactively test on the development train for future releases (e.g., Chrome Canary) to identify upcoming changes in WebRTC support.

Headsets

Headsets can improve audio quality by minimizing the echo possibility. We recommend the use of a headset on SnapCall Client calls to provide acoustic isolation between the speaker and microphone, and therefore reduce echo. That being said, headset selection should be made carefully.

PC Headsets: For lower-end PC hardware, we recommend USB Headsets over 3.5mm standard audio jack headsets. This allows you to bypass the native soundboard. For machines with a higher-end integrated soundboard, the 3.5mm connection should be fine.

Bluetooth Headsets: Bluetooth headsets can present unique challenges, as each headset operates slightly differently. If your headset uses a USB Bluetooth adapter, we recommend you pair it with the included adapter, rather than your device’s native Bluetooth receiver, to avoid interoperability issues.

Please note that Bluetooth support on mobile devices can vary significantly, and some devices may not provide the programmatic ability for Bluetooth answer/hang up buttons to be passed to non-native applications.

Static: Static and white noise issues with your client audio can often be due to a misbehaving or misconfigured headset. If you are experiencing static, try reproducing this issue with different headset hardware or no headset hardware to narrow down potential sources.

Device's OS

Microsoft Windows and Apple MacOS are the recommended operative systems to use Snapcall, though any device that is compatible with the browsers mentioned above can be used for the same purpose.

Mobile Devices: Hardware support, in general, is determined by the Mobile OS level supported by the hardware. Android Hardware, in particular, has a significant variation in behavior; even the same model can vary from region to region in areas such as AGC and Echo Cancellation behavior. If you’re experiencing a voice quality issue that you suspect to be due to mobile hardware, please perform the following tests if possible:

  • Test behavior with and without a headset

  • Test on different models of hardware

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