More and more businesses are using WLAN (Wireless Local Area Network) for their networks, and the advantages are quite obvious. WLAN allows devices to connect wirelessly, offering mobility and flexibility without the need for extensive cabling, aside from access points. Users today are much more mobile, and WLAN provides the freedom to move while staying connected. Multicell WLAN, which uses multiple access points to ensure seamless coverage over larger areas, is widely available and provides plenty of bandwidth for today's applications. In shared office spaces WLAN is often the only practical solution, allowing users to work efficiently, as just about any application works seamlessly with WLAN.

WLAN is also suitable for voice calls within these apps. For example, when a WhatsApp user calls another WhatsApp user, WLAN usually isn’t a problem. However, challenges arise when standard VoIP equipment is deployed.  

The Problem?

Several WLAN behaviors present challenges for VoIP calls. The first problem in a multicell network is that clients are constantly searching for a better signal. When a client decides to switch to another access point, it essentially holds back all the packets until the switch is completed, then it sends these packets in one burst. Depending on the configuration of the WLAN, this process can take several 100 ms, resulting in 20 or more RTP (Real-time Transport Protocol) packets arriving in bulk after a long time with no packets. RTP packets are used to deliver audio and video over IP networks, so this burst transmission can severely impact the quality of a VoIP call. 

Another problem is that WLAN tends to lose more packets than LAN. RTP uses UDP (User Datagram Protocol), which is a protocol designed for speed, but it doesn’t attempt to resend a packet when it’s lost. These packet losses often also occur in bursts, meaning multiple neighboring packets are lost together. This makes it increasingly difficult to reconstruct the content on the receiving end.

Most VoIP equipment has relatively short jitter buffers: devices want to keep the round trip delay short so the two participants in the conversation aren’t interrupting each other. While this approach works great for regular VoIP calls, it becomes a burden in WLAN environments because those bursts easily overwhelm the jitter buffer of most VoIP equipment. This leads to stuttering audio and can degrade the conversation quality to the point where communication becomes nearly impossible. 

Even when using applications like Microsoft Teams with standard VoIP equipment, users face the same problem. When calling from a laptop connected to a WLAN, the network creates the same kind of bursts with which other applications must contend. While calls from one Teams user to another Teams user can be managed more effectively by the large jitter buffers in Teams, most VoIP equipment will struggle to handle these bursts.

This doesn’t only affect VoIP devices. Most SIP trunks have the same issue, simply because they can’t know at what kind of endpoint the call ultimately terminates. This means, when calling from a WLAN device into a PSTN number, the quality of the call - from a WLAN network with large bursts - will be rather low. 

This creates frustration for users and administrators, since calls within the apps seem to be working fine.

The Solution

Fortunately, there are a few strategies to resolve these issues.  

The first step is to use codecs that are more robust against packet loss. OPUS is an audio codec designed for interactive real-time applications, and has become mainstream for many devices because it can handle packet loss quite well. Even when the VoIP equipment isn’t supporting OPUS, transcoding comes to the rescue and converts the audio stream into the right format. The loss in audio quality is usually negligible, compared to the quality with an older codec. While this approach addresses lost packets, it doesn’t fully resolve the problem of network packet bursts. The Vodia SBC supports OPUS and, by default, prefers it with WebRTC and apps.

The second step is to add a special network filter in the SBC (Session Border Controller) between WLAN devices and VoIP equipment, which dramatically increases the available jitter buffer for the conversation. Although this may introduce a longer round-trip delay, which can make conversations feel less immediate, it at least ensures that the audio is comprehensible. Most users are so used to it from their apps, so the number of complaints about this is quite low. Only VoIP experts would notice!

But how long should the burst jitter buffer be? One way lengthening the buffer is with static provisioning of a minimum jitter buffer length. At the beginning of the call, the SBC will hold back packets until the jitter buffer has reached a threshold value; after the threshold has been reached, packet playout can begin, significantly reducing the risk of running out of packets, even when there are bursts from the WLAN side. Although this may ensure that the audio is comprehensible, it will also add to the round-trip delay for calls that would otherwise sound fine. 

Alternatively, when using an adaptive filter, the jitter buffer only kicks in when a burst is detected during the call. While this leads to a gap in the conversation, it generally keeps round trip delay low and maintains higher call quality. Some apps actually try to learn in environments wherein they should start with a large jitter buffer. When you receive a prompt asking for call quality feedback after a conversation, that information may be used to set up the jitter buffer for future calls from the same IP address.

The SBC in the Vodia PBX is designed to detect bursts from WebRTC calls and, by default, inserts a jitter buffer with 1000 ms. This configuration works well in most cases, and administrators won’t have to worry about problems with the WLAN or having to explicitly program IP addresses to trigger the filtering.

As more and more companies adopt multicell WLAN with VoIP, it's become essential to have a smart SBC involved that takes care of transcoding and burst filtering. Ideally, system administrators aren't aware of these problems, and users can enjoy the same kind of experience with VoIP calls as they do with their mobile apps.

If you’re looking for the best cloud phone system, one that integrates seamlessly with your WLAN, look no further than the industry-standard Vodia PBX. For more information, contact us at sales@vodia.com or call +1 (617) 861-3490.

About Vodia Networks

Vodia Networks, Inc. is a pioneering provider of B2B Cloud Communications Solutions catering to enterprises, contact centers and service providers. Vodia's PBX software boasts an extensive suite of business telephony features for on-premise and cloud-based systems and operates seamlessly across Windows, Linux or Mac platforms. Fully compliant with SIP industry standards, the Vodia phone system integrates effortlessly with a wide range of SIP-based devices and trunking providers, granting ultimate freedom in telephony. Vodia’s multi-tenancy platforms are compatible with an unprecedented number of technologies, including desk phones, softphones and APIs, for myriad third-party software and CRM systems. Our mission is to empower our partners and end-users with the world's best cloud PBX and personalized support to ensure their success at every turn. Visit Vodia on LinkedIn, X and YouTube.