TNZI Tests 'Zero Hop' International Phone Calling Wednesday 16 January 2019 @ 09:10



Open RTP is touted as providing better voice call quality because the media is sent directly to the destination rather than passing through a number of companies to get there. This more direct route, with fewer hops, is said to provide an improved overall customer experience.

The signalling for those calls, to set up and end the calls, is however still carried in the traditional way and passes through the transit carrier. This does not affect the call quality as the signalling path is only used before and after the actual call.

Customers have been asking us for Open RTP routes, and suppliers have been providing Open RTP pricing to various destinations. We’ve noticed the frequency of being exposed to Open RTP is increasing. So being keen to learn about the potential to embrace Open RTP we first tested the configuration in our development lab and that worked fine. Our network systems can easily be configured. Then we jumped to a live concept trial with a partner in the USA for calls through to India. The trial is continuing and is providing valuable insights into how to fine-tune and adapt our back-end processes. It’s also giving us insights into how to possibly scale up the use of Open RTP to increase market penetration and improve customer experience in certain destinations.

Disruptive thinking

Open RTP is not a new technology - it is an enabler that changes the traditional business model of telecommunications companies (telcos) that have been in place for many decades. It disrupts traditional telco practise but leverages their relationships and connectivity for the call set-up and call ending (signalling). By ‘opening up’ the media stream (RTP) and allowing voice data to pass directly from source to destination, the path is more direct and has improved voice quality. It appears to be most useful where trust is perceived as an issue – due to the potential or actual high number of intermediaries that can be handling a voice call.

How it works

The SIP call flow differs in that the RTP (media stream) bypasses all of the telco hand-offs and flows directly from source (customer) IP address to destination (supplier) IP address. SIP signalling including INVITE, OK, ACK and BYE are all carried by the telco networks with no change. The exception is that the signalling passes the ‘direct’ IP addresses for the RTP stream rather than a telco IP address.

Service quality

While it’s all well and good to potentially improve voice quality to certain destinations by eliminating the number of hops that a voice call has to take, what about service quality? For example provisioning, billing, and incident management are parts of the overall service mix that provide either a good customer experience or not. We are looking at overall service quality as part of the trial and how best to fine-tune or business support systems and processes.

Game changer?

Time will tell but TNZI are interested to understand if Open RTP could change the nature of the traditional telco industry, maybe as VoIP did many years ago or is it simply an enabler to help penetrate high-volume low-decile markets?