Mastering WebRTC SDP: A Complete Guide to Analyzing Session Descriptions

0. Introduction

SDP (Session Description Protocol) is a historic format used in 20th‑century conferencing systems to exchange connection and media attributes between peers. Although JSON is now more common for RPC, SDP remains widely supported in traditional streaming devices, and understanding it is valuable for development and debugging.

1. SDP Specification Overview

SDP consists of a session‑level section followed by zero or more media‑level sections.

The session level starts with a v= line and ends before the first media line.

Each media section starts with an m= line and continues until the next media line or the end of the description.

Session‑level values serve as defaults for all media unless overridden.

An SDP description is a series of lines in the form <type>=<value>. The type is case‑sensitive; the value’s format depends on the type. Fields are separated by spaces, and complex lines may use semicolons ( ;) to separate parameters.

Key SDP fields include:

v= (protocol version)

o= (originator)

s= (session name)

t= (timing)

m= (media description)

c= (connection address)

b= (bandwidth)

a= (attributes)

k= (encryption key)

The fields must appear in a specific order: session description, time description, then media description.

2. WebRTC SDP Content Analysis

In WebRTC, the SDP is relatively flexible; it must contain at least the lines v o s t m c b a. A full example is shown below:

v=0<br/>o=mozilla...THIS_IS_SDPARTA-82.0.3 485627351772987711 0 IN IP4 0.0.0.0<br/>s=-<br/>t=0 0<br/>m=video 9 UDP/TLS/RTP/SAVPF 120 124 121 125 126 127 97 98<br/>c=IN IP4 0.0.0.0<br/>b=AS:1000<br/>a=sendrecv<br/>...

Media description example:

m=audio 9 UDP/TLS/RTP/SAVPF 111<br/>a=mid:audio<br/>a=rtpmap:111 opus/48000/2<br/>a=rtcp-fb:111 transport-cc<br/>a=fmtp:111 minptime=10;useinbandfec=1<br/><br/>m=video 9 UDP/TLS/RTP/SAVPF 96 97<br/>a=mid:video<br/>a=rtpmap:96 VP8/90000<br/>a=rtcp-fb:96 goog-remb<br/>a=rtcp-fb:96 transport-cc<br/>a=rtcp-fb:96 ccm fir<br/>a=rtcp-fb:96 nack<br/>a=rtcp-fb:96 nack pli<br/>a=rtpmap:97 rtx/90000<br/>a=fmtp:97 apt=96

Important format fields:

c=<nettype> <addrtype> <connection-address>

– connection info (IP address). m=<media> <port> <proto> <fmt> … – media type, transport protocol, and payload formats.

Other fields such as t= (start/stop time), s= (session name), and o= (originator) follow RFC definitions.

2.1 WebRTC‑SDP Introduction

WebRTC requires the lines v o s t m c b a. A complete example is provided above.

2.2 Plan B and Unified Plan

WebRTC SDP can follow two styles:

Plan B : Multiple audio/video streams are described within a single m= line.

Unified Plan : Each media track gets its own m= line, simplifying parsing.

With simulcast, a third level (track) is added, resulting in a three‑level hierarchy: session → media → track.

2.3 SDP Extension Information

The a= lines carry most of the detailed information, divided into media information (RTP parameters) and connection information (ICE, DTLS, etc.).

2.3.1 Codec Parameters

Examples:

a=rtpmap:108 H264/90000

a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f

a=rtcp-fb:108 goog-remb

The a=rtpmap line defines the payload type and codec; a=fmtp provides codec‑specific parameters; a=rtcp-fb lists supported RTCP feedback mechanisms (e.g., nack, pli, ccm‑fir, goog‑remb, transport‑cc).

2.3.2 Header Extension Parameters

Example:

a=extmap:14 urn:ietf:params:rtp-hdrext:toffset

2.3.3 Encoding Parameters

SSRC lines describe RTP streams:

a=ssrc:3245189021 cname:Cx4i/VTR51etgjT7<br/>a=ssrc:3245189021 msid:...<br/>a=ssrc:3245189021 mslabel:...<br/>a=ssrc:3245189021 label:...

Firefox typically uses a single a=ssrc line with a CNAME.

2.3.4 RTCP Parameters

a=rtcp:<port> IN IP4 <address>

– RTCP address (rare in WebRTC). a=rtcp-mux – RTP and RTCP share the same transport (RFC 5761). a=rtcp-rsize – reduced‑size RTCP packets (RFC 5506).

2.3.5 ICE Candidate

Examples:

a=candidate:0 1 UDP 2122252543 192.168.0.111 56774 typ host<br/>a=end-of-candidates

Candidate types include host, server‑reflexive (srflx), peer‑reflexive (prflx), and relay (TURN).

2.3.6 ICE Parameters

a=ice-options:trickle

– Trickle ICE. a=ice-lite – ICE lite mode. a=ice-ufrag:<value> and a=ice-pwd:<value> – ICE credentials.

2.3.7 DTLS Parameters

a=setup:actpass

– role negotiation (actpass/active/passive). a=fingerprint:sha-256 … – DTLS certificate fingerprint.

2.3.8 Miscellaneous

a=sendrecv

, a=sendonly, a=recvonly, a=inactive – media direction. a=group:BUNDLE 0 1 – bundle multiple media on one transport. a=mid:<id> – media identifier (RFC 5888). a=rid:<id> … – RTP stream identifier (draft‑rid). a=simulcast:send hi;mid;lo – simulcast configuration. a=bundle-only – indicates BUNDLE usage in answers. a=msid-semantic:WMS * – MediaStream identifier semantics.

References

[1] 2006, RFC4566 – SDP [2] WebRTC RTP Header Extension analysis [3] RFC5576 – SSRC Group [4] RFC5761 – RTCP‑Mux [5] RFC5506 – Reduced‑size RTCP [6] RFC8445 – ICE UDP [7] RFC6544 – ICE TCP [8] RFC5763 – DTLS‑SRTP [9] RFC5888 – Media Identification