WebRTC stream delivery requires particular audio and video codecs. In case the streams you are transporting via Zixi Broadcaster do not match these requirements, additional transcoding will be necessary in order to re-encode the stream to become WebRTC compliant.

The following are the requirements for a Zixi stream to be 

The following is a description of the possible methods for processing streams in Zixi for WebRTC viewing:

Single Version - Creating a single version of the monitor stream for all users

 – in this option the stream will be encoded using Zixi Broadcaster's transcoder (with existing X264 or Nvidia for transcoding). This option allows many users to monitor the stream without encoding for each individual user, which saves CPU capacity. However, the option will still require additional output bandwidth for each user. Since there is only one stream version, it is recommended to encode to a low resolution and low bitrate so that the stream will be viewable by all users monitoring the stream.

Multiple Version - Creating a unique version of the monitor stream for each user 

– in this option Zixi Broadcaster will transcode the video, transforming it into RAW frames, which will be passed to the WebRTC library, where it will be re-encoded for each individual user. In this option the stream will match the bitrate available for every particular user. Although this promises the best quality for users' specific network conditions, each re-encode consumes CPU, which could potentially max out the available CPU resources used to transport the streams. In this case, if the stream will be monitored by a large number of users, it is recommended to host a second Broadcaster and send the stream to the second Broadcaster and have all users monitor the stream from that second Broadcaster. By doing so, there will be no impact on the primary Broadcaster.

Video and Audio Codec Support Options

Creating Transcoding Profiles for WebRTC

Creating a WebRTC Output

Playing an input stream with WebRTC player

Configuring WebRTC Streams

(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. It’s designed to improve the performance of connection-oriented web appli

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 or s

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), which is widely used for sec

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).