Case Study: Inter-Protocol Transcoder

A Platform Under Pressure

The client is a UK telecommunications provider delivering mobile and unified communications services to business customers. Its VoIP proposition ran on a long-standing proprietary PBX platform that had proven reliable at scale, but only partially aligned with modern SIP expectations.

As customer usage matured, the operational importance of predictable, standards-compliant call behaviour grew. What had once been tolerable quirks became daily irritants - and increasingly, a support burden.

Where the Wheels Came Off

The provider's proprietary PBX remained stable and proven at scale. Mobile users had access to expected call features. From a platform perspective, nothing was failing.

Where the wheels came off was not in the core platform, but in the customer experience.

For customers using SIP desk phones, the service behaved differently than expected:

• Call transfers and conferencing worked on mobile clients - but were not available on SIP devices, creating inconsistency across endpoints.
• BLF presence did not exist on SIP phones, limiting visibility and coordinated call handling.
• A middle architectural layer prevented direct changes to the PBX, restricting straightforward remediation.
• As SIP adoption grew, so did the visibility of the gap - increasing support demand and commercial pressure.

The engine was sound. The strain was at the edge, where user expectations had outpaced what the architecture could deliver.

A Layer That Did the Heavy Lifting

Rather than forcing change on either end, we introduced an Inter-Protocol Transcoder as a middleware layer between SIP endpoints and the proprietary PBX.

Its purpose was simple but deliberate: translate SIP call control and presence events in real time, preserving SIP-native behaviour for users while allowing the core platform to continue speaking its own language. The result was functional parity at the edge, without destabilising the centre.

Designed for Those Who Rely on It

For end users, the change was immediate and practical. Call transfers, conferencing, and BLF presence behaved as expected, without workarounds or guesswork. SIP phones finally acted like SIP phones.

For operational teams, signalling mismatches were resolved at the point of interaction. Presence states aligned reliably with real call activity, reducing misroutes and avoidable interruptions.

For management, the solution delivered measurable results: fewer tickets, consistent feature execution, and predictable behaviour.

Introduced Without Drama

The transcoder was deployed to coexist with the existing platform, not replace it. By handling signalling and media efficiently, latency remained negligible and service reliability intact.

The open-source stack - Kamailio for SIP signalling, FreeSWITCH for media handling, with Lua and Node.js for custom logic - was selected for its proven performance in real-time voice environments and its ability to evolve alongside the business.

Results You Can Actually Measure

The operational impact was visible within weeks:

• 22% reduction in support tickets related to call control issues, driven by dependable transfer behaviour.
• 15% reduction in customer complaints linked to call disruption, following consistent BLF presence.
• 99.995% success rate for SIP feature execution under normal operating conditions.

A More Sustainable Way Forward

Beyond the metrics, the business gained confidence. Feature modernisation no longer implied platform upheaval. Support teams dealt with fewer repeat issues, supervisors faced fewer avoidable escalations, and leadership secured a controlled, incremental path to future evolution - on their own timetable.

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Client feedback has been paraphrased from direct discussions. Names, roles, and identifying details have been anonymised to protect confidentiality.