In the ever-evolving landscape of audiovisual communication, two protocols have long dominated the videoconferencing arena: H.323 and SIP (Session Initiation Protocol).
While the former marked the beginnings of IP-based videoconferencing, the latter has gradually established itself as the de facto standard. But where do things stand today? Let’s dive into the heart of this protocol duel.
H.323, standardized by the International Telecommunication Union (ITU), was one of the first protocols to standardize audio and video data transmission over IP networks. Its robustness and advanced security features made it a preferred choice for large enterprises and institutions requiring reliable and secure communications.
However, the complexity of its architecture and implementation has hindered its widespread adoption. Configuring firewalls and managing the numerous ports required for its operation could be tedious. Additionally, its lack of flexibility in the face of rapid IP network evolution has gradually limited its appeal.
SIP, on the other hand, is a lightweight, text-based signaling protocol standardized by the Internet Engineering Task Force (IETF). Its simplicity and flexibility have made it particularly well-suited for modern IP networks and integration with other web technologies.
The advantages of SIP are numerous:
Today, it’s clear that SIP has become the dominant protocol in the field of videoconferencing. The majority of new solutions, whether dedicated hardware systems, software platforms, or emerging technologies like WebRTC, rely on SIP to establish and manage communication sessions.
WebRTC = Real-time communication (audio/video/data) directly between browsers/applications, without plugins.
H.323 has not completely disappeared, however. It’s still found operating in older videoconferencing infrastructures, particularly within large organizations that have invested heavily in these systems.
Although SIP and H.323 are established standards for audio/video signaling, the major videoconferencing players often favor proprietary protocols optimized for their cloud infrastructures.
Zoom, for example, relies on its own multimedia transport protocols, secured by TLS and SRTP, for enhanced performance.
Microsoft Teams does not natively integrate SIP but ensures interoperability through certified SIP/H.323 gateways.
Google Meet also uses proprietary technologies based on WebRTC, offering limited compatibility with H.323/SIP through third-party services like Pexip.
Cisco Webex is a notable exception in maintaining native support for SIP and H.323, particularly for room systems.
This heterogeneity underscores the imperative of interoperability in an environment where traditional standards coexist with constantly evolving cloud solutions.
Although H.323 laid the foundations for IP videoconferencing, its complexity has gradually given way to the simplicity and flexibility of SIP, which has become the dominant pillar of current solutions. However, the emergence of proprietary protocols optimized by cloud giants is redefining the landscape, once again highlighting the crucial importance of interoperability to ensure smooth communication in an increasingly diverse ecosystem. The future of videoconferencing therefore lies in the ability to make these different approaches coexist, leveraging the strengths of each protocol to meet the specific needs of users.
In this context, the advent of artificial intelligence could also play a decisive role: by optimizing existing protocols, streamlining interoperability, or perhaps even giving rise to entirely rethought new solutions, relegating current protocols to the status of obsolete tools.