VoIP Quality Issues

VoIP, or Voice Over Internet Protocol, has some great advantages over plain analog voice networks. However, its quality always was the biggest problem one might face when using it. In early ages, the Internet was not adapted well for voice communications and most of the people has been uncertain whether to try it again since then. VoIP phones that had been promoted by many analysts as a future of communications did not justify hopes. The reasons included the poor quality, provided by low bitrate codecs, constant dropped calls and line congestions.

Here we'd like to mention the main causes of poor VoIP calls and how you can alleviate their influence and maximize quality.

1. Poor Internet

Perhaps, this is the main problem, which makes Internet calls unreliable and non pleasant. It always affects the quality so don't expect good calls if you have slow Internet connection or issues on the line.

You know, Internet providers usually don't care about voice traffic, they built their networks just for web surfing. Realtime traffic needs a high responsive connection that costs too much and not so profitable for providers.

There are recommendations for latency to be less than 150 ms one-way end-to-end for high-quality calls, but for international calls one-way delay up to 300 ms may be acceptable. Test your network before a call and contact the customer support if it seems the latency is out of reasonable limit.

There are still many people on the DSL Internet (ADSL, VDSL and so on), so if it is your current situation, call your provider and ask him to disable interleaving on your line. You will definitely feel the difference: test your connection before and after the interleaving disabling, you will notice the 10-40 ms drop in latency for sure. If your provider is unable to correct your network quality problems, you need to find a different provider. Changing providers is not a reliable solution, but sometimes it helps.

Our advice: choose your ISP wisely. Try to test it before or ask who already knows.

2. Equipment

It might surprise you, but your router can be a drawback too. Some of them are too slow to provide high-quality realtime traffic and sometimes halt for some time. IP packets delay rises, and many of them get dropped. It is very frustrating because the router is the last thing you would check if something goes wrong.

Many routers have Quality of Service (QoS) built-in feature. Get the router manual and turn it on for voice (SIP) traffic for smooth VoIP performance. Make sure it receives high priority so web surfing, YouTube videos watching or files downloading don't interrupt your important conversation. It is highly recommended both for enterprise and personal users regardless of their networks and equipment.

Our advice: set QoS feature on your router for voice traffic.

3. Connection method

Do you use Wi-Fi for your VoIP calls? It may be an issue too. Interference does not do good things, so it is always better to connect to an Ethernet when dialing somebody. However, often there is no option for traditional LAN. So you seriously you need to think about switching to 5Ghz Wi-Fi frequency and change access point's QoS settings. That can solve many problems, not only with VoIP calls alone. 2.4Ghz frequency is too overcrowded and sometimes causes network delays. You can ensure that is true by pinging a site while sitting on Wi-Fi: you will see pikes in latency from time to time.

Our advice: if you use Wi-Fi prefer 5Ghz frequency whatever you do. Don't forget about the separate QoS feature of your access point.

4. Inappropriate SIP settings

If you use a softphone or hardware SIP phone, spend some of your time to adjust your settings to get the most of the possible quality. Opt for the high-bitrate codecs always for the better quality. We, at Call2Friends, provide the best available G.711 mulaw codec by default. Make sure your choice is exactly like ours. It will deliver from unnecessary encoding/decoding processes which worsen quality and increase delays. However, bear in mind that the codec eats 80 kbit/s of traffic for one call and can be inappropriate for slow mobile networks.

Our advice: use G.711 mulaw codec whenever possible to minimize the quality loss.