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feat(webrtc-utils): move the offer SDP template alongside the answer #6572

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feat(webrtc-utils): move the offer SDP template alongside the answer
yexiyue Jul 28, 2026
a829b87
chore: fill in changelog PR numbers
yexiyue Jul 28, 2026
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4 changes: 4 additions & 0 deletions misc/webrtc-utils/CHANGELOG.md
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Original file line number Diff line number Diff line change
@@ -1,5 +1,9 @@
## 0.5.0

- Add `sdp::offer`, mirroring the existing `sdp::answer`. The offer template moved here
from `libp2p-webrtc`, where it was private.
See [PR 6572](https://github.com/libp2p/rust-libp2p/pull/6572).

- Revert migration to `quick-protobuf`, migrate back to `prost`.
See [PR 6363](https://github.com/libp2p/rust-libp2p/pull/6363).

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107 changes: 107 additions & 0 deletions misc/webrtc-utils/src/sdp.rs
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Original file line number Diff line number Diff line change
Expand Up @@ -26,6 +26,24 @@ use tinytemplate::TinyTemplate;

use crate::fingerprint::Fingerprint;

/// Renders the SDP offer, describing the client (the dialer).
///
/// The server does not verify the client's certificate -- the client's identity is established by
/// the Noise handshake that runs afterwards -- so a server rendering this offer on the client's
/// behalf passes [`Fingerprint::FF`] here.
pub fn offer(addr: SocketAddr, client_fingerprint: Fingerprint, client_ufrag: &str) -> String {
let offer = render_description(
CLIENT_SESSION_DESCRIPTION,
addr,
client_fingerprint,
client_ufrag,
);

tracing::trace!(%offer, "Created SDP offer");

offer
}

pub fn answer(addr: SocketAddr, server_fingerprint: Fingerprint, client_ufrag: &str) -> String {
let answer = render_description(
SERVER_SESSION_DESCRIPTION,
Expand All @@ -39,6 +57,95 @@ pub fn answer(addr: SocketAddr, server_fingerprint: Fingerprint, client_ufrag: &
answer
}

// An SDP message that constitutes the offer.
//
// Main RFC: <https://datatracker.ietf.org/doc/html/rfc8866>
// `sctp-port` and `max-message-size` attrs RFC: <https://datatracker.ietf.org/doc/html/rfc8841>
// `group` and `mid` attrs RFC: <https://datatracker.ietf.org/doc/html/rfc9143>
// `ice-ufrag`, `ice-pwd` and `ice-options` attrs RFC: <https://datatracker.ietf.org/doc/html/rfc8839>
// `setup` attr RFC: <https://datatracker.ietf.org/doc/html/rfc8122>
//
// Short description:
//
// v=<protocol-version> -> always 0
// o=<username> <sess-id> <sess-version> <nettype> <addrtype> <unicast-address>
//
// <username> identifies the creator of the SDP document. We are allowed to use dummy values
// (`-` and `0.0.0.0` as <addrtype>) to remain anonymous, which we do. Note that "IN" means
// "Internet".
//
// s=<session name>
//
// We are allowed to pass a dummy `-`.
//
// c=<nettype> <addrtype> <connection-address>
//
// Indicates the IP address of the remote.
// Note that "IN" means "Internet".
//
// t=<start-time> <stop-time>
//
// Start and end of the validity of the session. `0 0` means that the session never expires.
//
// m=<media> <port> <proto> <fmt> ...
//
// A `m=` line describes a request to establish a certain protocol. The protocol in this line
// (i.e. `TCP/DTLS/SCTP` or `UDP/DTLS/SCTP`) must always be the same as the one in the offer.
// We know that this is true because we tweak the offer to match the protocol. The `<fmt>`
// component must always be `webrtc-datachannel` for WebRTC.
// RFCs: 8839, 8866, 8841
//
// a=mid:<MID>
//
// Media ID - uniquely identifies this media stream (RFC9143).
//
// a=ice-options:ice2
//
// Indicates that we are complying with RFC8839 (as opposed to the legacy RFC5245).
//
// a=ice-ufrag:<ICE user>
// a=ice-pwd:<ICE password>
//
// ICE username and password, which are used for establishing and
// maintaining the ICE connection. (RFC8839)
// MUST match ones used by the answerer (server).
//
// a=fingerprint:sha-256 <fingerprint>
//
// Fingerprint of the certificate that the remote will use during the TLS
// handshake. (RFC8122)
//
// a=setup:actpass
//
// The endpoint that is the offerer MUST use the setup attribute value of setup:actpass and be
// prepared to receive a client_hello before it receives the answer.
//
// a=sctp-port:<value>
//
// The SCTP port (RFC8841)
// Note it's different from the "m=" line port value, which indicates the port of the
// underlying transport-layer protocol (UDP or TCP).
//
// a=max-message-size:<value>
//
// The maximum SCTP user message size (in bytes). (RFC8841)
const CLIENT_SESSION_DESCRIPTION: &str = "v=0
o=- 0 0 IN {ip_version} {target_ip}
s=-
c=IN {ip_version} {target_ip}
t=0 0

m=application {target_port} UDP/DTLS/SCTP webrtc-datachannel
a=mid:0
a=ice-options:ice2
a=ice-ufrag:{ufrag}
a=ice-pwd:{pwd}
a=fingerprint:{fingerprint_algorithm} {fingerprint_value}
a=setup:actpass
a=sctp-port:5000
a=max-message-size:16384
";

// See [`CLIENT_SESSION_DESCRIPTION`].
//
// a=ice-lite
Expand Down
4 changes: 4 additions & 0 deletions transports/webrtc/CHANGELOG.md
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Original file line number Diff line number Diff line change
@@ -1,5 +1,9 @@
## 0.10.0-alpha

- Move the offer SDP template into `libp2p-webrtc-utils` and render it through the new
`sdp::offer` there. No behaviour change.
See [PR 6572](https://github.com/libp2p/rust-libp2p/pull/6572).

- Update webrtc-rs to `v0.17` and fix libp2p noise data channel negotiation.
See [PR 6429](https://github.com/libp2p/rust-libp2p/pull/6429)

Expand Down
101 changes: 4 additions & 97 deletions transports/webrtc/src/tokio/sdp.rs
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Original file line number Diff line number Diff line change
Expand Up @@ -20,8 +20,8 @@

use std::net::SocketAddr;

use libp2p_webrtc_utils::Fingerprint;
pub(crate) use libp2p_webrtc_utils::sdp::random_ufrag;
use libp2p_webrtc_utils::{Fingerprint, sdp::render_description};
use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;

/// Creates the SDP answer used by the client.
Expand All @@ -42,103 +42,10 @@ pub(crate) fn answer(
///
/// Certificate verification is disabled which is why we hardcode a dummy fingerprint here.
pub(crate) fn offer(addr: SocketAddr, client_ufrag: &str) -> RTCSessionDescription {
let offer = render_description(
CLIENT_SESSION_DESCRIPTION,
RTCSessionDescription::offer(libp2p_webrtc_utils::sdp::offer(
addr,
Fingerprint::FF,
client_ufrag,
);

tracing::trace!(offer=%offer, "Created SDP offer");

RTCSessionDescription::offer(offer).unwrap()
))
.unwrap()
}

// An SDP message that constitutes the offer.
//
// Main RFC: <https://datatracker.ietf.org/doc/html/rfc8866>
// `sctp-port` and `max-message-size` attrs RFC: <https://datatracker.ietf.org/doc/html/rfc8841>
// `group` and `mid` attrs RFC: <https://datatracker.ietf.org/doc/html/rfc9143>
// `ice-ufrag`, `ice-pwd` and `ice-options` attrs RFC: <https://datatracker.ietf.org/doc/html/rfc8839>
// `setup` attr RFC: <https://datatracker.ietf.org/doc/html/rfc8122>
//
// Short description:
//
// v=<protocol-version> -> always 0
// o=<username> <sess-id> <sess-version> <nettype> <addrtype> <unicast-address>
//
// <username> identifies the creator of the SDP document. We are allowed to use dummy values
// (`-` and `0.0.0.0` as <addrtype>) to remain anonymous, which we do. Note that "IN" means
// "Internet".
//
// s=<session name>
//
// We are allowed to pass a dummy `-`.
//
// c=<nettype> <addrtype> <connection-address>
//
// Indicates the IP address of the remote.
// Note that "IN" means "Internet".
//
// t=<start-time> <stop-time>
//
// Start and end of the validity of the session. `0 0` means that the session never expires.
//
// m=<media> <port> <proto> <fmt> ...
//
// A `m=` line describes a request to establish a certain protocol. The protocol in this line
// (i.e. `TCP/DTLS/SCTP` or `UDP/DTLS/SCTP`) must always be the same as the one in the offer.
// We know that this is true because we tweak the offer to match the protocol. The `<fmt>`
// component must always be `webrtc-datachannel` for WebRTC.
// RFCs: 8839, 8866, 8841
//
// a=mid:<MID>
//
// Media ID - uniquely identifies this media stream (RFC9143).
//
// a=ice-options:ice2
//
// Indicates that we are complying with RFC8839 (as opposed to the legacy RFC5245).
//
// a=ice-ufrag:<ICE user>
// a=ice-pwd:<ICE password>
//
// ICE username and password, which are used for establishing and
// maintaining the ICE connection. (RFC8839)
// MUST match ones used by the answerer (server).
//
// a=fingerprint:sha-256 <fingerprint>
//
// Fingerprint of the certificate that the remote will use during the TLS
// handshake. (RFC8122)
//
// a=setup:actpass
//
// The endpoint that is the offerer MUST use the setup attribute value of setup:actpass and be
// prepared to receive a client_hello before it receives the answer.
//
// a=sctp-port:<value>
//
// The SCTP port (RFC8841)
// Note it's different from the "m=" line port value, which indicates the port of the
// underlying transport-layer protocol (UDP or TCP).
//
// a=max-message-size:<value>
//
// The maximum SCTP user message size (in bytes). (RFC8841)
const CLIENT_SESSION_DESCRIPTION: &str = "v=0
o=- 0 0 IN {ip_version} {target_ip}
s=-
c=IN {ip_version} {target_ip}
t=0 0

m=application {target_port} UDP/DTLS/SCTP webrtc-datachannel
a=mid:0
a=ice-options:ice2
a=ice-ufrag:{ufrag}
a=ice-pwd:{pwd}
a=fingerprint:{fingerprint_algorithm} {fingerprint_value}
a=setup:actpass
a=sctp-port:5000
a=max-message-size:16384
";