While recently wrapping up a project with a large number of devices connected over wireless, we encountered some challenges with the quality of the audio connection. These sorts of issues are not uncommon, particularly within the realm of Voice over Internet Protocol (VoIP) applications. The key difference between VoIP and many other web applications is its real-time nature; unlike traditional Web applications, wherein data can be buffered and processed before delivery to the user, VoIP requires the transmission and reception of audio data in real-time to ensure seamless communication. This real-time requirement introduces unique challenges in maintaining audio quality, as factors such as jitter, network latency and packet loss often result in negative impacts on the user experience. Additionally, the quality of the audio connection often depends on the type of codec used, the efficiency of the audio processing algorithms and the hardware components involved in the reception and transmission of audio data.

Finding the needle in the haystack

Our experience investigating media-related issues during a past project involving a large PCAP file with concurrent calls prompted us to develop a feature for our PBX that records the packets associated with each call, along with the call record. This feature greatly simplified the process of identifying and troubleshooting media issues, including bursts, excessive jitter or one-way audio.

The ability to conveniently find a problematic call in the call log and download the corresponding packets saves a significant amount of time. Encrypted media presented a challenge, however, as playback of the actual audio in Wireshark was complex and time-consuming. To address this, our phone system now records the decrypted RTP packets for easy playback, eliminating the need for manual extraction and decryption of SDES crypto lines from SDP in INVITE packets. Unfortunately today, for DTLS-encrypted media streams, there isn’t a way to decrypt the media stream after the PCAP has been recorded, which highlights the effectiveness of DTLS in securing media streams.

Further optimization involved enriching the packet information with details such as destination IP addresses, source and timestamps. In Linux, we leverage the socket's ability to report the actual timestamp when a packet is received, along with the IP addresses. While Windows only provides the IP addresses, it’s still an improvement over the original PCAP recording, where the addresses had to be inferred by the PBX.

Our ongoing efforts to refine the troubleshooting tool have greatly improved accuracy and efficiency. By recording decrypted RTP packets, capturing additional packet information and leveraging operating system capabilities, we have streamlined the process of identifying and resolving media-related issues. This enables a seamless, high-quality communication experience for our users.

Why not for every call?

The cost of hard drive space has decreased significantly, making it affordable to store large amounts of data. Calls typically occupy only a few MB, which is less space than modern camera photos require. Additionally, the CPU overhead associated with storing call data is minimal.

Privacy concerns arise, however, when considering call recording: calls contain sensitive information, and recording them carries various obligations and liabilities. For instance, callers may need to be notified their calls may be recorded, which is impractical in typical PBX deployments during media problem troubleshooting.

To address these concerns, we have introduced two additional options in the PBX. The first option allows for the inclusion of SIP packets in every call record - SIP packets typically amount to a few KB of data, so even if the PBX records millions of calls, the additional hard drive space required is an insignificant amount for modern servers. Moreover, since SIP packets contain metadata about calls, the compliance requirements are much lower, as compared to actual conversation data. The caller ID is already included in the CDR, and the technical details of the call add limited information from a privacy perspective. This option is enabled by default, but it can be disabled by the system administrator.

The second option replaces the PBX media with dummy data. While every media packet is still added to the packet trace, the actual content is erased, ensuring there is no way to playback the actual content. Although playback is no longer possible, all other information is retained; this facilitates the identification of typical issues such as one-way audio, packet drops and burst problems.

Time is money

Everybody knows installing and operating hundreds of access points isn’t possible without management tools capable of measuring the performance of the access points and the registered clients, the cost for such a project would explode. While home WLAN access points can be operated without such tools, complexities arise when multiple cells come into play. The same is true for VoIP projects that go beyond a handful of extensions. Tools and data collection are crucial for a successful project within a limited time budget. 

VoIP platforms like the Vodia PBX can integrate with external tools like VoIP monitors by sharing TLS master keys - this enables the decryption of packets. Other tools like sipcapture focus on SIP signaling, which provides valuable insights. When it comes to delving deeper into the timing of media packets, however, these tools fall short. This is where the Vodia PBX steps in and provides the media packets as well, encrypted or not, with meaningful data that can be analyzed by standard tools.

Most VoIP projects are executed at a professional level - they need to be profitable and stay within budget. Having the right data and the proper tools for analysis is paramount in achieving these goals. The PBX collects relevant data and makes it accessible for these tools to facilitate the identification and resolution of issues.

The future holds even more promise, thanks to the potential integration of Artificial Intelligence (AI) in data analysis: AI could provide suggestions or even make changes to the setup based on the collected data. Such tools are still in their nascent stages, and human intelligence remains indispensable for successful deployment in complex environments.

Having the right tools and data is essential for troubleshooting and optimizing telecommunication systems, especially in enterprise deployments. As technology advances, we can expect even more sophisticated tools and techniques to emerge, further enhancing our ability to manage and maintain these critical systems effectively and efficiently.

We are glad to help our partners

Vodia’s expertise in enterprise VoIP business makes it so our partners can deploy the highest quality professional services for impeccable customized communication experiences. Don’t hesitate to contact us with any inquiries, sales@vodia.com, (617) 861-3490.

About Vodia Networks

Vodia Networks, Inc. is a pioneering provider of B2B Cloud Communications Solutions catering to enterprises, contact centers and service providers. Vodia's PBX software boasts an extensive suite of business telephony features for on-premise and cloud-based systems and operates seamlessly across Windows, Linux or Mac platforms. Fully compliant with SIP industry standards, the Vodia phone system integrates effortlessly with a wide range of SIP-based devices and trunking providers, granting ultimate freedom in telephony. Vodia’s multi-tenancy platforms are compatible with an unprecedented number of technologies, including desk phones, softphones and APIs, for myriad third-party software and CRM systems. Our mission is to empower our partners and end-users with the world's best cloud PBX and personalized support to ensure their success at every turn. Visit Vodia on LinkedIn, Twitter and YouTube.