Clear voice communication allows your agents to take care of customers quickly and accurately. If agents and customers experience dropped calls, distortion, and choppiness, then everyone is frustrated and unhappy.
The internet is not designed for real-time applications like VoIP. As traffic passes through the internet, latency, jitter, and lost packets cause voice distortion and choppiness on the call. This makes communication difficult. Here are a few of the problems that can happen during a call over the internet.
Latency is the time it takes for a data packet to reach its destination and return a response. If latency is too high during a VoIP call, the connection can be choppy or lost.
Data packets are expected to be received at regular intervals. Jitter is the condition where the spacing is inconsistent. Some packets are received close together, while long gaps occur between others. This can cause distortion in VoIP calls.
Data packets can be lost during transmission. This is usually caused by long queue times on the network or by misdirection. Most VoIP applications try to fill in the blanks, but if too many packets are lost, the compensation is overwhelmed, degrading call quality.
Subspace has redesigned the internet for today’s real-time applications. By creating our own active mesh optical transport network overlaying the internet, we became the only global platform routing traffic at the maximum speed in real-time.
We offer high-performance network optimization for multiplayer games, modern fintech solutions, ecommerce apps, and real-time voice and video applications.
We use a proprietary combination of internet weather-mapping, custom networking stacks, omniscient routers, and dedicated fiber. We partner with ISPs around the world to bring software publishers and telecommunications companies like you the quality and speed your users demand.
We’re optimized for real-time protocols. Whether you need to accelerate TCP/UDP traffic with our PacketAccelerator, move your SIP media faster than ever before with SIPTeleport, deploy global TURN without switching on a server with GlobalTURN, or remove dependency on SBC with a single API call on RTPSpeed, we’ve got you covered.
We use a proprietary AI to find the fastest route across the network for your data. This means that when your application is running in Subspace, it always has the lowest latency possible.
Access Subspace with a single API call and have the same great network performance anywhere in the world, no matter where your customers or agents are located.
The SIPteleport proprietary algorithms ensure that your real-time application delivers consistent audio, free of stutters, jitters, or latency, under even the most challenging network conditions, with up to 80% reduction in packet loss.
With the rise in popularity of online gaming and the demand for an enhanced online experience, we sought a partner that could help open up opportunities to improve gameplay for those in traditionally poor-performing regions. Subspace understood our digital vision and helped to improve latency and guarantee the best possible gaming experience for their users. Subspace is revolutionizing the user experience across the globe, and we are thrilled to be in partnership with them.
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