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T2 · T2-R06 · SIP, WebRTC, RTP

RTP, SRTP, and WebRTC

Three layers, again. SIP or the browser signaling sets the session up. SDP describes it. RTP carries it, and RTCP reports on it. SRTP is the secure form named by RFC 3711. WebRTC is the browser suite. RFC 8825 tells you which specifications belong in that suite. It is not a codec list and not a vendor SDK.

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Counts on this page

4boxes
5opened sources
1confirm rows
5study lines

Opened sources

Confirm rows, no URL

Learn it in 5 minutes

  • Wikipedia says RTP delivers audio and video over IP, typically over UDP, and that RTCP monitors transmission statistics and quality of service.
  • The same article says RFC 3550 in 2003 superseded RFC 1889, and that RTP is often used with a signaling protocol such as SIP.
  • RFC 3711 is the Secure Real-time Transport Protocol, SRTP.
  • RFC 8825 is an overview of real-time protocols for browser-based applications. The abstract says the document is an applicability statement and does not itself specify a protocol.
  • Wikipedia says WebRTC lets audio and video work inside web pages by direct peer-to-peer communication, and that ICE, STUN, and TURN are the NAT traversal techniques it uses.

Visual

RTP, SRTP, and WebRTC Media. Teaching sketch. Confirm a live system. RTP RTCP SRTP WebRTC Names come from pages opened in 2026-10. A missing name stays confirm.

Count the boxes: 4. Each label is a name from a page or a study label marked on the drawing.

Flow

RTP, SRTP, and WebRTC Caller SIP Called session setup session setup RTP, not required to hairpin Teaching sketch from RFC 3261 and RFC 3550. Not a packet capture.

What is going on

Three layers, again. SIP or the browser signaling sets the session up. SDP describes it. RTP carries it, and RTCP reports on it. SRTP is the secure form named by RFC 3711. WebRTC is the browser suite. RFC 8825 tells you which specifications belong in that suite. It is not a codec list and not a vendor SDK.

ICE, STUN, and TURN are names on the WebRTC article. This session did not open their separate RFCs, so a port number or a timeout for them stays confirm.

Worked example

A browser call fails only for some home routers. Write WebRTC, then ICE, STUN, and TURN as the article's traversal names. Write confirm beside any port you did not read. If the call is a SIP phone with one-way audio, start from RTP and RTCP instead.

Quiz

1. RFC 8825 says it does not itself specify a protocol.

2. RTCP is the protocol that carries the voice samples.

Ecosystem

Sister pieces sit on the ecosystem page: the glossary, the mind map, the integration map, and the bridges.

  • RTP — The media transport Wikipedia describes for audio and video, usually over UDP, specified in the RFC 3550 lineage.
  • WebRTC — The browser real-time project Wikipedia describes. RFC 8825 is the IETF overview of the protocol suite, not a single protocol.

Config. Write signaling, description, and media as three lines. Add SRTP only when the design says the media is protected.

Troubleshooting. Signaling success plus silence points at RTP, SRTP keys, or the NAT traversal names. Do not restart by rewriting the dial plan.

Bridge. LiveKit's docs, later, call their transport WebRTC. That product page is not RFC 8825. Cite both when you join them.

Mind map, sister technologies, and future hooks live on the ecosystem page.

Sources and go deeper

Opened 2026-10. A row marked confirm has no link, because that page was not verified.

sources.json Markdown