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docs(rutster-media): complete rutster-trunk rename in rtc_session doc

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The RtcSession module doc cited the dropped rutster-signaling-sip crate
and a 'future SIP SDP path' that ADR-0007 superseded. Re-anchor on
ADR-0007: no first-party SIP/SDP — the trunk is rented (CPaaS media-leg)
or out-of-tree SBC, so rutster's SDP is WebRTC-only via str0m. Final
straggler of the rutster-signaling-sip → rutster-trunk rename.

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01KhhqKG4cra7d1PBoVbj9UJ
This commit is contained in:
opencode controller
2026-06-30 01:24:41 -04:00
parent 1a029d7044
commit 9128fd5d0a
1 changed files with 6 additions and 5 deletions
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11
crates/rutster-media/src/rtc_session.rs
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@@ -10,11 +10,12 @@
//! str0m 0.21's `Rtc::sdp_api().accept_offer(offer)` produces the SDP
//! answer natively: DTLS fingerprint (from the cert str0m generates), ICE
//! ufrag/pwd, and codec negotiation (Opus, the only codec we registered).
//! Slice 1 does NOT hand-roll an SDP munger — str0m's path is the spec's
//! "embryo of the future SIP SDP path" (§3.7). When step 5 brings SIP/SDP
//! negotiation into `rutster-signaling-sip`, that crate may extract shared
//! SDP helpers from str0m or build its own. Slice 1's WebRTC-ICE-coupled
//! SDP lives entirely in str0m.
//! Slice 1 does NOT hand-roll an SDP munger — str0m handles WebRTC
//! offer/answer natively. There is no first-party SIP/SDP path: under
//! [ADR-0007](../../../docs/adr/0007-trunk-rented-transport.md) the trunk is
//! rented (a CPaaS media-leg fork — raw audio, no SDP) or an out-of-tree SBC
//! (which owns any carrier SIP/SDP, outside the trust boundary). rutster's
//! WebRTC-ICE-coupled SDP lives entirely in str0m.
use std::net::SocketAddr;
use std::time::{Duration, Instant};