Why Improving VoIP Needs More Than Optimized SD-WAN and SIP


Too many businesses are running real-time voice and video apps on an internet that was never meant to support them–resulting in spotty communications that don’t live up to the promise of next-gen networks like SD-WAN. With GlobalTURN by Subspace, get the true potential of SD-WANs through optimized routing that delivers fast, clear connectivity.

It’s hard to believe, but it’s been nearly 20 years since the “can you hear me now?” guy went viral and captured the frustration of poor-quality voice and video calls.

Even more unimaginable? Two decades later, too many users of voice apps are still asking that same annoying question when their calls don’t perform as expected.

Despite the technological leaps we’ve made to improve network connectivity, voice lags and dropped calls are still a problem when it comes to real-time communication. But often, the issue isn’t with the apps themselves. More commonly, despite their many benefits, ‘next-gen’ networks are not optimized for providing clear, crisp, and reliable voice connectivity.

Software-Defined Wide-Area Networks (SD-WANs) are often hyped as the “ideal” solutions for quality of service and reliability. But when it comes to supporting communications apps, they still have some WebRTC limitations. If we want to improve real-time performance and eliminate jitter and packet loss issues, it’s time to think beyond SD-WANs and prevent these common issues with routing that’s always on — and always optimized.

SD-WAN: Just One Piece of the High-Performance Connection Puzzle

With SD-WANs, administrators can more efficiently scale bandwidth up and down based on demand while also gaining end-to-end network visibility and the ability to manage their networks via a centralized SDN controller. This helps to improve network agility, connectivity, IT costs, and other management burdens.

But despite these benefits, SD-WANs aren’t the be-all-and-end-all some may expect them to be, especially where voice applications are concerned.

For starters, the quality of service can vary tremendously depending on your vendor. Second, implementing an SD-WAN can be a complex and costly undertaking — which can compound frustrations when call quality issues do occur. Another significant limitation to be aware of is that while SD-WANs do make it easier to connect to the cloud, they can’t fix other problems that might plague calls after you’re connected. When it comes to VoIP, issues like jitter and latency can still occur — and the only thing an SD-WAN can really do to improve them is to try another route.

If your network needs to support real-time communications, the most significant limitation to be aware of is that SD-WANs on their own can’t do all the work. To optimize WebRTC, you will still need to layer on additional solutions, called Traversal Using Relays around NAT (TURN) servers.

How TURN Servers Work — and Why They Increase Real-Time Protocol Limitations

The exchange works well for some protocols, but it breaks down for VoIP.

The main reason for these WebRTC limitations on SD-WAN is that NATs route traffic based on the information contained in IP headers. This is sufficient for protocols such as HTTP and POP. But VoIP carries important connectivity information within its IP packets, where the NAT can’t access it.

This is where TURNs come in. In order to get around the real-time protocol limitations that are inherent to NATs, TURNs route traffic through a relay server. This may help facilitate the connection, but it also introduces the potential for call quality issues — specifically buffering delays and hairpinning if the routing is through a relatively distant server. Also less than ideal: the TURN server must be available for the entire call duration.

Because of these limitations, telcos needed a better solution, leading them to interactive connectivity establishment (ICE), which uses multiple TURN servers to find the fastest path.

Improve Real-Time Performance with Always On, Always Optimized Routing

But — good news — this doesn’t put us back where we started. There’s an even better, faster way.

With Subspace’s GlobalTURN solution, path selection is always on and always optimized. Instead of routing traffic through a single relay point, GlobalTURN is distributed, reducing bottlenecks. Meanwhile, the fact that GlobalTURN utilizes a direct path prevents issues such as hairpinning and hops. At the same time, GlobalTURN protects clients’ IP addresses, so there’s also no need to choose between voice app performance and privacy.

For SD-WANs, this means finally no more dial and wait and no more annoying call quality issues after you’re connected. Instead, you get a clear, low lag, instant connection at the push of a button, resolving the inherent and oh-so-frustrating WebRTC limitations that exist with SD-WAN.

Conclusion: Fast, High-Quality Voice and Video — At the Push of a Button

And sadly, many of the most common solutions to these limitations don’t fully bridge the gaps.

At Subspace, we’re delivering on the true potential of SD-WANs, with optimized routing that delivers fast, clear connectivity. Finally, no more delays and interruptions when it comes to making calls. With GlobalTURN, all you have to do is connect.

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