Mobile Real-Time Communication (MRTC) is an mPaaS component that provides feature-rich audio and video calls. It supports one-to-one calls and multi-party conferences on various terminals, such as PCs, mobile devices, and Internet of Things (IoT) devices. During a call, you can use features such as screen recording, screen sharing, and screenshots. The component also supports instant text messaging, file transfers, and real-time speech recognition to detect voice confirmations and determine a user's intent. The video-on-demand feature lets you play videos, presentations, and other on-screen content during a video call.
MRTC provides high security with end-to-end encryption that complies with China's national cryptographic standards. All video recording, transmission, and storage processes are encrypted.
Features
Video calls
Multiple participation modes: Supports one-to-one and multi-party video calls.
Multiple platforms: Supports iOS, Android, PC web, H5, and miniapps.
Multi-terminal interconnection: Supports interconnection between mobile phones, PCs, and IoT devices.
Session persistence: Maintains session continuity during temporary network issues or network transitions to ensure that service flows are not interrupted.
Custom and adaptive video specifications: Supports custom width, height, maximum frame rate, and maximum bitrate. The video specifications automatically adjust based on network conditions within the specified limits.
Call recording
Granular recording control: Supports audio-only, video-only, or mixed audio and video recording. You can record a single participant, two participants, or multiple participants.
Multi-end recording: Supports recording on the client-side or the server-side based on your security requirements.
Multiple layout modes: Supports Picture-in-Picture (PiP) and nine-square grid modes.
Watermark overlay: Supports real-time overlays of text, image watermarks, and timestamps for tamper-proofing.
Multiple storage systems: Supports NAS, OSS, AFTS, and local file systems.
Multiple formats: Supports common formats, such as FLV and MP4.
Data communication
Instant messaging: Supports text messaging for communication before a video call.
File transfer: Allows you to upload a file and send the link to other participants for download.
Command pass-through: Provides a transparent and reliable command transmission channel for your services.
Automation and intelligence
Automatic voice risk alerts: Automatically broadcasts risk alerts and other information to replace manual instructions.
Video on demand: Supports playing risk alert content, such as videos and presentations, during a video call.
Automatic transcription: Transcribes video call audio in real time to create a text record of the conversation that is organized by time and speaker.
Automatic confirmation: Detects a customer's voice confirmation for risk alerts to help agents determine the customer's intent.
Enhanced features
Screen sharing: Allows you to share your local screen content with other participants.
Screenshots: Supports remote camera control for taking photos and capturing local screenshots.
System forwarding
Bypass live streaming: Supports cloud-based audio mixing, video composition, and forwarding to a live streaming system. You can ingest streams to Alibaba Cloud CDN and third-party live streaming platforms.
Benefits
High security
Dynamic signing and signature verification.
End-to-end encryption across the entire link that complies with China's national cryptographic standards.
Dynamic key authentication for OpenAPI.
Encrypted transmission and storage of recordings.
Adaptive video quality in weak network conditions
Accurately evaluates effective network bandwidth based on packet loss type, packet loss rate, response time (RT), and network jitter.
An optimized video quality adaptation algorithm adjusts video definition, smoothness, and size in real time based on the target bitrate.
Prioritizes video quality in the region of interest.
Highly reliable media stream and signaling transmission
A real-time streaming media platform ensures stable transmission.
Dynamic switching between redundant and optimized links.
A two-layer, highly reliable signaling mechanism provides both node-to-node and end-to-end reliability. This mechanism reduces latency and improves reliability to 99.95%.
Low latency
Ultra-low latency, reaching 200 ms for remote control applications.
Fast focus.
An improved Forward Error Correction (FEC) algorithm that tolerates up to 40% packet loss.
H.264 High Profile technology reduces bandwidth consumption by up to 50%.
Multi-terminal access
Supports access from multiple terminals, such as PCs, mobile phones, tablets, miniapps, and IoT devices. Allows interconnection between different terminals and apps.
Rich features
Dual recording: Cloud-based bidirectional recording with multiple granularities, formats, custom watermarks, and storage methods.
Intelligent agent: Automatic voice risk alerts, video on demand, and automatic transcription.
Simple APIs and easy integration
A lightweight version is available for easy integration.
Fast service integration: Integrate the service in as little as one week from development to launch.
Service architecture
The following describes the service architecture for audio and video calls:
The system architecture for audio and video calls is as follows:
Audio and video SDK: A client software development kit (SDK) that supports Android, iOS, PC web, miniapps, Linux, and Windows.
Signaling service: Manages signaling connections, forwards signals, manages channels, and maintains the session lifecycle.
master: A load-balancing server that provides balanced scheduling and liveness management for the media server.
rtcmcu: A media forwarding service that receives, forwards, processes, and maintains the status of media streams.
rtcrecorder: A recording service that records and stores calls.
rtcvod: A video-on-demand service that provides on-demand video and audio streaming.