The excitement that is Cisco Live! took to the stage in Las Vegas recently, allowing the company to show off its brand-new intent-based networking system. Of course, the event also gives Cisco's technology partners a chance to strut their stuff too - including ThousandEyes, who demonstrated their network monitoring service, and their new UC management capabilities, designed to offer an insight into /unified-communications/ucaas deployments.
While there are plenty of management tools out there that can give you a glimpse into the world of UC performance when the solution is on-premise, ThousandEyes' solution is one of the first options to offer visibility all the way to cloud services, and for hybrid solutions too. Last week, the company announced the availability of SIP server and Voice Call tests to provide end-to-end visibility in UCaaS and VoIP performance.
Full Visibility into UCaaS Performance
Back in the days of time-division multiplexing, voice ran all by itself on a dedicated endpoint network, making troubleshooting a more straightforward solution for everyone involved. Today, modern solutions run on VMs (Virtual Machines), and clients move around, with the network shared among a range of people. This all makes problem-solving pretty challenging. If you add the issue of moving call control to the cloud, where most businesses have no control of visibility, and troubleshooting what's going wrong gets even harder.
Most of the existing solutions for voice monitoring are reactive, based on call detail records and packet capture. Using these techniques, network managers would analyse issues after a problem is found. That's far too late to find the source of the issue. Additionally, the tools don't allow for capacity planning or benchmarking. They provide little insight into actual real service deliver because they don't have correlation of data within the application, routing, and network layers.
ThousandEyes Visibility for VoIP Calls
The Voice Call and SIP Server tests from ThousandEyes are the new solutions added to the company's existing RTP portfolio tests, designed to offer complete visibility across all stages of VoIP calls. SIP uses a complex process to establish calls, in which each endpoint needs to complete various tasks, such as the registration of a call server. Once the call is in place, the endpoints can connect with each other using the real-time transport protocol.
