WebRTC (Web Real-Time Communication) is a free, open-source project that provides web browsers and mobile applications with real-time communication (RTC) via simple application programming interfaces (APIs). It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps. Supported by Apple, Google, Microsoft, Mozilla, and Opera, WebRTC is being standardized through the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).

About WebRTC

Video and Audio Codec Support Options

Creating Transcoding Profiles for WebRTC

Creating a WebRTC Output

Playing an Input Stream with WebRTC Player

Configuring Sources for WebRTC

Viewing Sources with WebRTC

WebRTC for ZEN Master

How to Monitor Streams Using WebRTC

ECN lets a congested router mark a packet to signal congestion rather than dropping it. Sender marks its packets as ECN-capable; a congested router along the path can then flip the bits to signal congestion without packet loss.

(Decoding Time Stamp) a timestamp embedded in MPEG transport streams and other video container formats that indicates when a frame should be decoded by the decoder. Needed for In advanced codecs - frames can arrive out of order because of B-frames

(Presentation Time Stamp) indicates the exact time when a video or audio frame should be presented during playback. It ensures synchronization between audio and video streams.

(Program Map Table) Describes all elementary streams (video, audio, metadata) for a specific program, along with their PID numbers; Found on the PID indicated by the PAT

(Program Association Table) Lists all programs in the transport stream and points to their corresponding PMT; Always carried on PID 0x0000

Time Addressable Media is a set of principles from BBC Research & Development (BBC R&D) which set out a practical approach for how to handle broadcast content throughout the media lifecycle from live to packaged content.

TAMS

(Differentiated Services Code Point) a 6-bit field within the IP header of a network packet. The 6-bit value (ranging from 0 to 63) classifies the type of traffic and its desired priority level.

DSCP

Refers to a transport stream packet that is 204 bytes in length, which is a variation of the standard 188-byte TS packet.

204B_TS_Packet

(Program Clock Reference) helps synchronize the decoder's clock with the encoder's clock. This synchronization is essential for maintaining audio-video sync and ensuring smooth playback.

(Estimated Peak Signal-to-Noise Ratio) a variation of the traditional PSNR metric, developed to estimate video quality without needing access to the original (reference) video—which is a key limitation of standard PSNR

ePSNR

(Peak Signal-to-Noise Ratio) a widely used metric in image and video processing to measure the quality of a reconstructed or compressed image compared to the original.

PSNR

(Common Media Application Format) a media streaming format developed by Apple and Microsoft and standardized by the MPEG group to simplify and unify the delivery of streaming video across different platforms and devices.

CMAF

(Quick UDP Internet Connections) a modern transport layer network protocol developed by Google and later adopted by the IETF (Internet Engineering Task Force) as a standard designed to improve performance.

QUIC

DTS-PCR delta refers to the time difference between the Decoding Time Stamp (DTS) and the Program Clock Reference (PCR). Understanding this delta is important for ensuring proper synchronization and smooth playback of audio and video streams.

DTS-PCR_delta

(Real-Time Streaming Protocol) a network control protocol used to control streaming media servers. It allows clients to remotely control a media stream—similar to how you use a remote control for a video player

RTSP

(Source-Specific Multicast) SSM is a method of IP multicast where receivers specify both the multicast group address they want to receive traffic from and the specific source address (i.e., the sender) they want to receive traffic from

(Real-time Transport Protocol) a protocol used in media streaming, especially for real-time audio and video transmission over IP networks

A network management technique used to control the flow and volume of data being transmitted over a network. Its main goal is to optimize performance, reduce latency, and ensure a smooth streaming experience, especially when bandwidth is limited.

Traffic_Shaping

(Datagram Transport Layer Security) a security protocol designed to provide privacy, integrity, and authentication for datagram-based applications. It is essentially the datagram version of TLS (Transport Layer Security).

DTLS

(Network Interface Card) a hardware component—either built into a computer's motherboard or added as an expansion card—that allows a device to connect to a network, such as a local area network (LAN) or the internet.

(Constant Bitrate) a method of encoding audio or video where the bitrate remains the same throughout the entire stream. This means that the amount of data used per second is fixed, regardless of the complexity of the content being streamed.

(Signal Processing Notification): The XML message that is returned from the SDS outlining any stream switches or SCTE35 manipulation that should occur.

(Signal Processing Event): The XML message that is sent to the SDS when triggered by a relevant SCTE message

(Signal Decision Service): Used in broadcast and streaming workflows to manage ad insertion, content switching, or emergency alerts based on predefined rules

the Event Scheduling and Notification Interface (ESNI) standard, designed to help broadcasters and streaming platforms manage content distribution dynamically

SCTE-224

A standard for digital program insertion in advertising systems. It defines interfaces and protocols that allow broadcasters and streaming platforms to manage ad placement

SCTE-130

A standard for signaling ad insertion and content switching in digital video streams. It defines a way to embed cue messages within a transport stream

SCTE-35

(Society of Cable Telecommunications Engineers): A professional organization that focuses on advancing broadband and cable technology through standards development

SCTE

A component of SCTE-130 that manages available ad slots in video streams

POIS

(Event Scheduling and Notification Interface): Part of the SCTE 224 standard, which helps manage event scheduling, blackout enforcement, and content replacement in IP-based video delivery

ESNI

(Event Signaling and Management): A protocol used in video streaming to refine and condition SCTE 35 signals within a live stream

ESAM

A one-to-one stream delivery method where each receiver gets a unique copy of the stream

Unicast

(Multiple Video Programming Distributor):  A service provider that delivers multiple television channels to consumers

MVPD

A method of broadcasting video over IP where one stream is sent to many receivers without duplicating the stream per recipient

Multicast

(Digital Multiple Video Programming Distributor): A term used to describe services that provide multiple TV channels over the internet

DMVPD

(Content Delivery Network): A network of servers that distributes and caches content geographically to reduce latency and improve delivery

(Adaptive Bitrate Streaming): A technique in streaming where the video quality is adjusted in real time based on network conditions and device performance

The process of converting a video file or stream from one encoding format or bitrate to another

Transcoding

TR_101

(Root Cause Analysis): Analysis of problems anywhere in the ingest, processing, and distribution of streams to determine the exact underlying cause.

(Packet Identifier):  A 13-bit field in an MPEG transport stream (MPEG-TS) that helps receivers distinguish between different types of data packets

(Over-The-Top): Delivery of video content directly to consumers via the internet, bypassing traditional cable or broadcast systems

(Maximum Transmission Unit):  The largest size of a data packet that can be transmitted over a network without fragmentation

device or software that compresses raw video/audio into a specific format for distribution (e.g., H.264, HEVC)

Encoder

counterpart to an encoder; it decompresses video/audio streams for playback or further processing

Decoder

Automatic Repeat Query - protocols responsible for automatic retransmission of packets that are found to be corrupted or lost during the transmission process

The Standard from the Society of Motion Picture and Television Engineers that describes how to send digital video over and IP networ

SMPTE_2022

Network bonding is a process of combining or joining two or more network interfaces together into a single interface

Bonding

Baseband is considered the near zero compression of video being routed around the broadcaster facility

Baseband

(Asynchronous Serial Interface): a method for transmitting MPEG Transport Streams (MPEG-TS) over coaxial cable or optical fiber. Used in broadcasting and video distribution to transport compressed digital video and audio signals between equipment

(Web Real-Time Communication): A real-time communication protocol that enables direct, low-latency audio, video, and data transmission between browsers and devices without plugins

WebRTC

(User Datagram Protocol): A connectionless transport protocol that transmits data without error correction or retransmission, prioritizing speed and low latency

(Secure Reliable Transport): An open-source transport protocol similar to Zixi, designed for low-latency, secure video delivery over IP

(Real-Time Messaging Protocol): A streaming protocol originally developed by Adobe for low-latency video, audio, and data transmission, commonly used to ingest live video streams into CDNs

RTMP

(Reliable Internet Stream Transport): An open standard streaming protocol built on UDP that adds features like error recovery, packet retransmission, and encryption to enable reliable, low-latency video transport over lossy or public IP networks

RIST

(Transport Stream): A standard container format for transmission and storage of audio, video, and data, widely used in broadcasting.

MPEG-TS

(MPEG-1 Audio Layer III): A digital audio compression format that reduces file size while maintaining sound quality.

(HTTP Live Streaming): An adaptive bitrate streaming protocol developed by Apple that delivers video over standard HTTP connections by breaking content into small chunks

High Efficiency Video Codec - video compression standard designed to improve efficiency over its predecessor, H.264 (AVC). Offers 25% to 50% better compression, making it ideal for streaming, broadcasting, and video formats like 4K and 8K

HEVC

a video compression standard designed to improve efficiency over its predecessor, H.264 (AVC). It offers 25% to 50% better compression while maintaining the same level of video quality

H.265

(also known as Advanced Video Coding [AVC]): a widely used video compression standard designed to deliver high-quality video at lower bit rates

H.264

(Dynamic Adaptive Streaming over HTTP): An MPEG-standardized adaptive streaming protocol that delivers video over HTTP by dynamically switching bitrates based on network conditions

DASH

(Advanced Audio Coding): A widely used audio compression format that provides better sound quality than MP3 at similar bit rates

A proprietary protocol developed by Zixi for reliable, low-latency, error-corrected transport of live video over IP networks.

Zixi_Protocol

A gateway server that can receive and redistribute streams to multiple endpoints, supporting protocol conversion, transcoding, and routing.

Zixi_Broadcaster

the live video orchestration and telemetry control plane that enables Zixi users to manage large-scale configuration and monitoring of the Zixi Enabled Network, Zixi’s live streaming platform, devices and appliances.

ZEN_Master

(Zixi-as-a-Service): ZaaS is a cloud-based offering of Zixi’s platform in which Zixi provides and manages the Zixi Broadcasters, allowing the customer to focus entirely on managing content and distribution.

ZaaS

A Zixi app for remote contribution from mobile devices

OnAir

(Loss Of Redundancy): Loss of backup systems designed to ensure continuous playback and fault tolerance

Settings within Zixi that balance between ultra-low latency and resiliency, depending on network conditions

Latency_Mode

Interruptible Foldback Support: IFB or switched talkback is a monitoring and cueing system for one-way communication allowing the decoder to send audio back to the encoder with a Zixi ZEC or Feeder. IFB support hast been implemented in Zixi Broadcast

A feature in Zixi that allows seamless switching between redundant feeds for undisrupted streaming without packet loss or viewer interruption

Hitless_Failover

The FEC Block Latency (ms) parameter sets the number of milliseconds that Zixi protects with a single FEC block

FEC_Block_Latency

(Forward Error Correction): A method of error correction that adds redundant data to help recover lost packets without needing retransmission

This indicator detects cases where the source encoder configuration is creating excessively lossy content. The analysis does not rely on the bitrate, but rather extracts the global quantization parameter, which is a parameter that indicates how much 

I-Frame_Quantization_Parameter

Detects frozen video, i.e. no motion, for a duration of about 60s or longer

Frozen_video_detection

Detects blank frames of any color for a duration of about 60s or longer

Blank_video_detection

Detects silence (dBFS< -90) on AAC or MPEG audio with a duration of about 30s or longer.

Audio_silence_detection

Detects audio volume level for every channel on AAC or MPEG audio. The analysis of the various audio tracks is available in the audio tree. The volume is presented in dBFS (Decibels relative to full scale).