WebRTC integration

WebRTC is a technology for real-time communications right in the browser: audio and video calls, voice/video rooms, streaming, P2P data and file transfer — without plugins. The connection can be directly between users (peer-to-peer) or via a media server. Honestly about complexity, this is key: 'just a video call in the browser' is actually a complex system. A signaling server (to establish the connection), STUN/TURN servers (for NAT/firewall traversal — without TURN some connections will not establish, and TURN traffic is paid), handling network drops and reconnections, echo/noise cancellation, quality adaptation to the channel are needed. It is serious engineering work, not 'turn on the camera'. Honestly about scaling (P2P is not for crowds): pure P2P works great for 1-on-1 and small groups, but for many participants it does not scale (each sends a stream to each — load grows quadratically). For group calls/webinars a media server (SFU/MCU) is needed — separate infrastructure, capacity and cost. We honestly assess the number of participants and choose the architecture. Honestly about 'no quality guarantee': call quality strongly depends on the users' own networks (speed, loss, mobile internet). WebRTC adapts, but perfect connection on a poor user network cannot be guaranteed. We make it robust but are honest about the limit. Honestly about excess: for tasks that do not need genuine real-time media (e.g. asynchronous video, record and watch later) WebRTC is excessive — ordinary upload/streaming is simpler. We will honestly assess whether WebRTC specifically is needed. Honestly about the effect: for genuine call/media tasks it gives real-time right in the browser, but it is complex infrastructure justified by a specific need. Honestly about access: a genuine real-time task, signaling/TURN infrastructure, a team are needed. An important boundary: this is WebRTC (P2P/media real-time); WebSocket (real-time data) — 1061; streaming recorded content — adjacent. Picture this: instead of 'calls via third-party services or not at all' — your own real-time media in the browser where it is genuinely needed. The base price starts from 80,000 ₽ (depends on the scenario and number of participants).

• Video/audio calls or P2P exchange right in the browser are needed.
• Third-party communication services do not fit (own/embedded is needed).
• You want WebRTC but underestimate the complexity (TURN, scale, network).
• Group calls on pure P2P cannot handle the load.

• WebRTC integration (calls/media/data, P2P or via a media server)
• A signaling server, STUN/TURN for NAT traversal, reconnections
• Choosing the architecture by the number of participants (P2P vs SFU/media server)
• Quality adaptation, drop handling and fallbacks
• Honest boundaries (complex; P2P not for crowds; no quality guarantee on a poor network; excessive for non-real-time)
• A link with WebSocket (1061) for signaling/data
• Accounting for TURN traffic and infrastructure costs
• Handover and review with you

• Real-time audio/video/data right in the browser (where needed)
• An architecture for the real number of participants (P2P or media server)
• Robustness: NAT traversal, reconnections, quality adaptation
• Honest boundaries (complex infrastructure; no network-quality guarantee; not for non-real-time)

• We assess the scenario and number of participants (is WebRTC needed and which architecture)
• We implement signaling, STUN/TURN, the media stream, fallbacks
• We account for costs and the quality limit, honestly set boundaries with you

• Is WebRTC just turning on video in the browser?

  No, honestly: behind a 'simple video call' is a complex system — a signaling server, STUN/TURN for NAT traversal (without TURN some connections will not establish, and TURN traffic is paid), handling drops and reconnections, echo/noise cancellation, quality adaptation. It is serious engineering work and infrastructure, not 'turn on the camera'. We honestly build in this complexity.

• Will WebRTC handle group calls/webinars with many participants?

  On pure P2P — no, honestly: P2P works great for 1-on-1 and small groups, but for many participants the load grows quadratically and does not scale. For group calls a media server (SFU/MCU) is needed — separate infrastructure, capacity and cost. We honestly assess the number of participants and choose the architecture rather than promise 'P2P for everyone'.

• Will the connection be of perfect quality?

  It cannot be guaranteed, honestly: quality strongly depends on the users' own networks (speed, loss, mobile internet). WebRTC adapts quality to the channel and works robustly, but perfect connection on a poor user network is impossible to ensure — it is beyond our control. We make it as robust as possible (adaptation, reconnections), honestly indicating the limit: we do not control the user's network quality.