WHEN enables adding browser based low latency viewing of streaming content.

It is an IETF draft specification (here), which adds a simple signaling layer on top of WebRTC that can be used to connect a WebRTC publisher to a WebRTC media server – usually streaming services and/or CDNs.

The player in this case only receives media and doesn’t send media.

Since WebRTC doesn’t include its own signaling protocol, WHIP has been proposed, for solving a very specific market need – in the streaming industry, there is a clear decoupling between the various network components – publishers, media servers and players. Each application or service provider can mix and match these three entities as he sees fit.

For WebRTC to fit into the streaming industry, such decoupling was needed, and as a result, there was a need for a clear signaling protocol to connect the media server to the player, which is what WHEP does.

See also WHIP

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WHIP enables connecting browser based users and live inputs directly to streaming services.

It is an IETF draft specification (here), which adds a simple signaling layer on top of WebRTC that can be used to connect a WebRTC publisher to a WebRTC media server – usually streaming services and/or CDNs.

The publisher in this case only sends media and doesn’t receive media.

Since WebRTC doesn’t include its own signaling protocol, WHIP has been proposed, for solving a very specific market need – in the streaming industry, there is a clear decoupling between the various network components – publishers, media servers and players. Each application or service provider can mix and match these three entities as he sees fit.

For WebRTC to fit into the streaming industry, such decoupling was needed, and as a result, there was a need for a clear signaling protocol to connect the publisher to the media server, which is what WHIP does.

See also WHEP

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MQTT is a common protocol used in Internet of Things use cases.

At times, MQTT is used with WebRTC applications. The biggest example is Facebook Messenger, which makes use of MQTT and its voice and video calling features are built using WebRTC.

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Signaling multiple media sources is needed to be able to send a video along with screen sharing, multiple camera sources or routing multiple streams from an SFU to the users.

In Unified Plan, there is an m= line per each media source.

The mechanism chosen for the official WebRTC specification is Unified Plan. The competing approach is known as Plan B.

Today, all browsers support it by default.

Signaling Multiple Media Sources

In multiparty real-time communication setup (most group calling solutions), the ability to handle multiple media sources is important.

This includes the capability to send a video alongside screen sharing, managing multiple camera sources, or routing multiple streams from a Selective Forwarding Unit (SFU) to the users.

The necessity for a standardized plan arose from the need to ensure seamless interaction among these media elements, and thus, the Unified Plan was conceived.

Structure

The core structure is defined by the presence of an m= line for each media source within the SDP.

This m= line is a fundamental aspect as it delineates each media source, thereby facilitating the organized signaling of multiple media streams.

The structuring is what sets Unified Plan apart and aligns it with the official WebRTC specification.

Adoption and Browser Support

Unified Plan, over time, has garnered widespread adoption and has become the mechanism of choice as per the official WebRTC specification, superseding the competing (and not deprecated or unimplemented) approach known as Plan B.

The transition was driven by its robust structure and the ease it introduced in managing multiple media sources.

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Signaling multiple media sources is needed to be able to send a video along with screen sharing, multiple camera sources or routing multiple streams from an SFU to the users.

In Plan B, an m= line is an “envelope” that includes multiple media sources per one defined transport.

The mechanism chosen for the official WebRTC specification isn’t Plan B but rather Unified Plan.

Plan B has been deprecated and removed from browsers.

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SDP munging can take place before passing the local SDP created using createOffer() and/or createAnswer() to setLocalDescription(), or before passing the SDP received over the network to setRemoteDescription().

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JSEP is an IETF draft describing the offer/answer architecture that WebRTC uses to connect sessions using SDP.

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IMS is the architectural framework defined for delivery of IP services for the telecom industry. It uses SIP as its signaling protocol.

While IMS doesn’t directly relate to WebRTC, IMS vendors are adding WebRTC support and capabilities into their product portfolio; mainly as an additional access point into their network.

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XMPP is a protocol used mainly for presence and instant messaging. An XMPP extension called Jingle adds the ability to initiate voice and video calls as well.

XMPP can be used as a signaling protocol in WebRTC.

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SIP is a widely used signaling protocol in VoIP that is used a lot in telecom.

To adapt to WebRTC, SIP introduced SIP over WebSocket. This enables running a SIP session via a web browser where the media is handled by WebRTC.

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