71 lines
2.0 KiB
Rust
71 lines
2.0 KiB
Rust
//! Modules for events in the `m.call` namespace.
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//!
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//! This module also contains types shared by events in its child namespaces.
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pub mod answer;
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pub mod candidates;
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pub mod hangup;
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pub mod invite;
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#[cfg(feature = "unstable-msc3401")]
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pub mod member;
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pub mod negotiate;
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pub mod reject;
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pub mod select_answer;
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use serde::{Deserialize, Serialize};
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/// A VoIP session description.
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///
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/// This is the same type as WebRTC's [`RTCSessionDescriptionInit`].
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///
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/// [`RTCSessionDescriptionInit`]: (https://www.w3.org/TR/webrtc/#dom-rtcsessiondescriptioninit):
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#[derive(Clone, Debug, Deserialize, Serialize)]
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#[cfg_attr(not(feature = "unstable-exhaustive-types"), non_exhaustive)]
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pub struct SessionDescription {
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/// The type of session description.
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///
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/// This is the `type` field of `RTCSessionDescriptionInit`.
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#[serde(rename = "type")]
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pub session_type: String,
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/// The SDP text of the session description.
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///
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/// Defaults to an empty string.
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#[serde(default)]
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pub sdp: String,
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}
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impl SessionDescription {
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/// Creates a new `SessionDescription` with the given session type and SDP text.
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pub fn new(session_type: String, sdp: String) -> Self {
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Self { session_type, sdp }
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}
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}
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/// The capabilities of a client in a VoIP call.
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#[cfg(feature = "unstable-msc2747")]
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#[derive(Clone, Debug, Default, Serialize, Deserialize)]
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#[cfg_attr(not(feature = "unstable-exhaustive-types"), non_exhaustive)]
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pub struct CallCapabilities {
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/// Whether this client supports [DTMF].
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///
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/// Defaults to `false`.
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///
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/// [DTMF]: https://w3c.github.io/webrtc-pc/#peer-to-peer-dtmf
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#[serde(rename = "m.call.dtmf", default)]
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pub dtmf: bool,
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}
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#[cfg(feature = "unstable-msc2747")]
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impl CallCapabilities {
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/// Creates a default `CallCapabilities`.
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pub fn new() -> Self {
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Self::default()
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}
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/// Whether this `CallCapabilities` only contains default values.
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pub fn is_default(&self) -> bool {
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!self.dtmf
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}
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}
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