Trickle ICE

SDK-managed WebRTC uses Trickle ICE by default. This applies to initial offers and answers, re-INVITEs, and ICE restarts; there is no enablement option.

For inbound SDK-managed calls, the Web Phone reads p-rc-ice-servers from the INVITE before constructing the peer connection. A present list replaces registration-provided STUN servers for that Call Session; an absent header keeps the registration configuration. A malformed present header makes answer() reject and leaves the Call Session ringing. Delegated WebRTC is unchanged. The header supplies server configuration, not candidates: browser WebRTC gathers candidates from those servers, and later candidates continue through the existing SIP INFO path.

For initial outbound SDK-managed calls, the INVITE carries the generated offer before it is applied locally. The first matching provisional response with a valid server list configures the peer connection and starts gathering. Provisional responses without the header remain nonterminal. If none supplies a list, the final response starts gathering with its list or the registration STUN fallback. The saved offer is applied once; this does not create another offer, ICE restart, or re-INVITE. A malformed present header fails call setup. Delegated WebRTC keeps its existing offer and gathering contract.

Candidate ownership boundary

The Web Phone owns SIP signaling, dialog routing, candidate ordering, and ICE generation lifetime. It sends the generated offer or answer SDP immediately, keeping the candidates already present in it: a candidate that is part of the sent SDP is not signaled again by INFO. Later local candidates are sent in gathering order, one SIP INFO request at a time, followed by end-of-candidates. If a candidate INFO request is rejected or receives a non-success response, the Web Phone stops that ICE generation; the Call Session remains active and no fallback re-INVITE is started.

Incoming SDP fragments are converted once into browser-shaped RTCIceCandidateInit values, preserving each candidate value, and forwarded in wire order. Candidates are held in a small buffer until the remote description is ready and are then applied in arrival order. The SDK validates nothing about ordinary candidates: browser WebRTC validates its candidate input, and a delegated application owns its addRemoteCandidate implementation. The SDK checks only an incoming null end marker, ignoring it when its username fragment does not match the active remote ICE generation. Incoming parsing and application failures are nonfatal to the Call Session; later valid candidates continue.

When a re-INVITE or ICE restart starts a newer ICE generation, or the Call Session is disposed, the queued candidate work of superseded generations is dropped.

Delegated WebRTC

A delegated WebRtcSession opts in by exposing the complete optional trickleIce capability:

trickleIce: {
  setLocalCandidateHandler(
    handler: (candidate: RTCIceCandidateInit | null) => void,
  ): void;
  addRemoteCandidate(candidate: RTCIceCandidateInit | null): Promise<void>;
}

The application registers and emits browser-shaped local candidates through setLocalCandidateHandler; it emits one null end-of-candidates marker per ICE generation. Emitted candidates are forwarded as received, so candidate validation stays with the application. Incoming candidates are passed to addRemoteCandidate for the application to apply. The Web Phone owns SIP construction, dialog routing, candidate ordering, and generation lifetime.

An opted-in session's offer and answer SDP must contain one audio media section and a=ice-options:trickle. The Web Phone rejects missing advertisements rather than rewriting application-owned SDP. A delegated session without the complete capability retains the existing complete-SDP behavior.

Supported scope

This is the Trickle ICE subset supported by RingCentral Web Phone audio calls. It does not claim complete RFC 8840 interoperability or support for arbitrary SIP servers.