LEARNING-BASED AUTOMATED CALL FLOW TESTING

Invented by Bharrat; Shaun Jaikarran, Chaudhary; Sanchit, Nayak; Subhransu S., Subramanian; Rajangam, Chand; Subhashis, Ouellet; André, Ramachandran; Gopannan
Automated testing is changing how we check if our network equipment works well, especially for voice calls over the internet. This article dives deep into a new patent that introduces a smart, learning-based way to test SIP call processing devices—like Session Border Controllers (SBCs)—with real-world data. You’ll discover why this matters, how it works, and what makes it stand out.
Background and Market Context
In the world of communication networks, devices like Session Border Controllers (SBCs) make sure that calls using the Session Initiation Protocol (SIP) reach the right place and stay secure. These devices are found in almost every company or service provider that allows people to make phone calls over the internet.
But there is a big problem. Whenever there is a new feature or a new software release for these devices, companies need to test everything before putting it in their real, working networks. If something goes wrong with a call, it can mean dropped connections, unhappy customers, and lost money. So, testing is important. But it is also slow, expensive, and often misses real-world problems.
Most tests are done by people, by hand, or with simple scripts. This takes a lot of time. It is also easy to miss rare or weird call flows that only happen on a live network with real users. Because of this, many businesses wait a long time before updating their systems, even if there are important new features or security fixes. They just don’t want to risk breaking things.
This slow process means companies spend too much money on testing and not enough on making new things. They also face security risks by not updating quickly. Even worse, when they do test, they often only check if a call connects, not if everything in the call works as it should. Real-world calls can be very different and much more complicated than what is tested in the lab.
There is a clear need for a better way to test SIP devices. The method should be automatic, fast, and should make sure that actual, real-life call flows are tested. It should be able to capture what really happens on a live network and use that as the basis for testing new versions or features.
This patent introduces a solution that does exactly that. It uses real call records from active networks to learn what typical and unusual call flows look like. Then, it builds automatic tests based on this data and compares how new devices or software handle these same call flows. This kind of learning-based testing is a big step forward for the whole industry.
Scientific Rationale and Prior Art
To understand why this new approach works so well, let’s look at how things have been done up to now, and where they fall short.
Traditionally, SIP device testing has relied on manually written test cases. These test cases try to cover as many situations as possible, but they are only as good as what the test designer can imagine. Often, the tests are based on simple call flows—like placing a call and then hanging up. But real networks are messy. Calls can be forwarded, transferred, recorded, or dropped in all sorts of ways.
Manual tests also struggle to keep up with changes in the network. If the actual, live network configuration changes (new devices are added, routing changes, etc.), the lab setup may not match anymore. This makes lab tests less useful. If a rare problem happens on the live network, it may never be tested in the lab, and it may reappear after a software update.
Some previous solutions have tried to automate parts of testing. For example, you can use tools to generate fake SIP traffic. But these tools still need someone to tell them what messages to send and what to check for. They don’t “learn” from real data, and they don’t adapt as the network changes.
Other approaches have tried to capture traffic from live networks in the form of packet captures (PCAPs) and replay them in a lab. While this can help, it is hard to manage and doesn’t easily create test cases that match different network setups. There’s also the challenge of privacy and data security—real calls may contain sensitive information.
The key problem is that real call flows are dynamic. New call patterns appear, and old ones fade away. Manual and even partially automated methods can’t keep up with this constant change. Also, they often lack detailed checks; they may only look for call completion, not whether all the signaling and headers are correct.
What sets the patented approach apart is its use of continuous, real-world data sampling. Instead of relying on guessed scenarios, it samples a small percentage of live calls—just enough to capture a full picture over time without gathering too much data. It then learns the unique “shape” and details of each call flow, down to the individual SIP messages, headers, and parameters.
Another important difference: the system automatically updates lab configurations to match production ones, using a process called “twinning.” This ensures that the lab device is as close as possible to the real one, so tests are meaningful. It can also anonymize data, protecting privacy.
Finally, the system doesn’t just check if a call completes. It checks the sequence and content of every message, header, and parameter, comparing them carefully against what happened in the live network. This makes it much more likely to catch subtle bugs or changes in behavior.
In summary, the new method overcomes the main problems of the old ways: it keeps up with real-world changes, it is automatic, and it checks much more thoroughly.
Invention Description and Key Innovations
Let’s look at how the patented system works, step by step, and why it is so powerful.
1. Learning from Real Calls
The system starts by collecting call detail records (CDRs) and trace records (TRCs) from a production SIP device, like an SBC, running in a live network. It does not need to collect every call—just a small, random sample over time. This sample is enough to eventually see almost every kind of call flow that the device handles.
Each sample includes detailed information about the call: which messages were sent and received, what headers were present, the order of events, and other important features. The system then processes these records to extract a “call flow record.” This is like a fingerprint for each kind of call. It includes:
– The sequence of SIP messages (like INVITE, 100 TRYING, 180 RINGING, 200 OK, BYE, etc.)
– The headers and parameters in each message
– Which trunks or routes were used
– Any special things, like call recording or encryption
If two calls have the same features, they are treated as the same “call flow.” The system prunes duplicates, so it only keeps unique call flows for testing.
2. Building a “Twinning” Lab Setup
It is not enough to just know what calls happen in the real network. The lab device used for testing must also be set up to match the real one, so the test makes sense. This is done using a process called “production twinning.” The system downloads the configuration from the real device and automatically transforms it for the lab. For example, it changes IP addresses, ports, and other settings so the lab device can work in the test environment, but everything else matches.
The system can create a list of “transforms” that map production addresses and settings to lab ones. It can even reuse existing mappings when possible, reducing human work. If any settings can’t be mapped automatically, a person can fill in the blanks.
This twinning process ensures that the lab device behaves just like the real one. If the production network changes, the twinning process can be run again to update the lab.
3. Generating Automated Test Cases
Once the unique call flows are known and the lab device is set up, the system automatically creates test cases. For each call flow, it generates a test script (for example, using the popular open-source SIPp tool) that simulates the exact sequence of SIP messages, headers, and parameters seen in production.
The test scripts are smart—they can mimic any endpoints, servers, or network elements involved in the call. They also use the lab-specific addresses created during twinning, so everything lines up.
These test cases are stored and can be run any time, for example, after a software update or when a new feature is added.
4. Running and Auditing Tests
When it’s time to test, the system loads the correct configuration onto the lab device and runs the test script. During the test, it collects new CDRs and trace records for each test call.
The system then generates a “test case call flow record” from these new records, just like it did for the original call flow. Now comes the critical step: it compares the test case call flow record with the original reference call flow record.
This comparison checks:
– Are all the SIP messages present, in the right order?
– Do all the headers and parameters match?
– Are any extra or missing?
– Are things like call routing, recording, or encryption handled the same way?
If anything is different, the test fails. The system can even score each mismatch by severity and keep track of recurring issues. If a feature is non-deterministic (like random routing), it can retry the test several times to see if the difference is just due to randomness, not an actual bug.
5. Reporting and Continuous Improvement
The system creates audit reports showing which tests passed or failed, which features matched or did not, and details about any errors. These reports can be visualized in dashboards or exported for further analysis.
Because the whole process is automated and based on live data, it can be run regularly—catching problems before they reach customers. It can also be customized for each customer’s unique network and traffic patterns.
Key Innovations
– Learning from real, live network data: Instead of relying on guesses, the system builds a model of actual call flows using real CDRs and traces.
– Automatic configuration “twinning”: The lab device is set up to match production, with minimal human effort, making tests meaningful and accurate.
– Unique, detailed call flow records: Tests cover not just simple call completion, but all signaling details, headers, and parameters, catching subtle bugs.
– Automated test generation and execution: Test cases are built and run automatically, saving time and money.
– Comprehensive auditing and reporting: Results are checked at a fine level of detail and reported clearly, making it easy to pinpoint issues.
– Adaptable and scalable: The system can handle any network, adapt to changes, and scale to large numbers of test cases.
This learning-based automated testing method is a game-changer for anyone who manages or develops SIP devices. It makes testing faster, smarter, and more complete. It helps companies deliver new features sooner, keep their networks secure, and avoid costly bugs.
Conclusion
Testing SIP devices is hard, but it doesn’t have to be. By using real-world data, automating the setup, and checking every detail, the patented system described here makes it easy to spot problems before they reach users. It’s a practical, proven way to keep networks reliable and secure, while letting businesses move faster and spend less on manual work. If you are responsible for SIP device testing, adopting this approach can help you deliver better results, more quickly, and with greater confidence.
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