active-call 0.3.54

A SIP/WebRTC voice agent
Documentation
use crate::{CallOption, ReferOption, media::recorder::RecorderOption, synthesis::SynthesisOption};
use serde::{Deserialize, Serialize};
use serde_with::skip_serializing_none;
use std::{
    collections::HashMap,
    sync::{Arc, Mutex},
};

pub mod active_call;
pub mod sip;
pub use active_call::ActiveCall;
pub use active_call::ActiveCallRef;
pub use active_call::ActiveCallType;

pub type CommandSender = tokio::sync::broadcast::Sender<Command>;
pub type CommandReceiver = tokio::sync::broadcast::Receiver<Command>;

// WebSocket Commands
#[skip_serializing_none]
#[derive(Debug, Deserialize, Serialize, Clone)]
#[serde(
    tag = "command",
    rename_all = "camelCase",
    rename_all_fields = "camelCase"
)]
pub enum Command {
    Invite {
        option: CallOption,
    },
    Accept {
        option: CallOption,
    },
    Reject {
        reason: String,
        code: Option<u32>,
    },
    Ringing {
        recorder: Option<RecorderOption>,
        early_media: Option<bool>,
        ringtone: Option<String>,
    },
    Tts {
        text: String,
        speaker: Option<String>,
        /// If the play_id is the same, it will not interrupt the previous playback
        play_id: Option<String>,
        /// If auto_hangup is true, it means the call will be hung up automatically after the TTS playback is finished
        auto_hangup: Option<bool>,
        /// If streaming is true, it means the input text is streaming text,
        /// and end_of_stream needs to be used to determine if it's finished,
        /// equivalent to LLM's streaming output to TTS synthesis
        streaming: Option<bool>,
        /// If end_of_stream is true, it means the input text is finished
        end_of_stream: Option<bool>,
        option: Option<SynthesisOption>,
        wait_input_timeout: Option<u32>,
        /// if true, the text is base64 encoded pcm samples
        base64: Option<bool>,
        /// Customizing cache key for TTS Result
        cache_key: Option<String>,
    },
    Play {
        url: String,
        play_id: Option<String>,
        auto_hangup: Option<bool>,
        wait_input_timeout: Option<u32>,
    },
    Interrupt {
        graceful: Option<bool>,
        fade_out_ms: Option<u32>,
    },
    Pause {},
    Resume {},
    Hangup {
        reason: Option<String>,
        initiator: Option<String>,
        headers: Option<HashMap<String, String>>,
    },
    Refer {
        caller: String,
        /// aor of the calee, e.g., sip:bob@restsend.com
        callee: String,
        options: Option<ReferOption>,
    },
    Mute {
        track_id: Option<String>,
    },
    Unmute {
        track_id: Option<String>,
    },
    History {
        speaker: String,
        text: String,
    },
    Custom {
        sender: Option<String>,
        data: serde_json::Value,
    },
}

/// Routing state for managing stateful load balancing
#[derive(Debug)]
pub struct RoutingState {
    /// Round-robin counters for each destination group
    round_robin_counters: Arc<Mutex<HashMap<String, usize>>>,
}

impl Default for RoutingState {
    fn default() -> Self {
        Self::new()
    }
}

impl RoutingState {
    pub fn new() -> Self {
        Self {
            round_robin_counters: Arc::new(Mutex::new(HashMap::new())),
        }
    }

    /// Get the next trunk index for round-robin selection
    pub fn next_round_robin_index(&self, destination_key: &str, trunk_count: usize) -> usize {
        if trunk_count == 0 {
            return 0;
        }

        let mut counters = self.round_robin_counters.lock().unwrap();
        let counter = counters
            .entry(destination_key.to_string())
            .or_insert_with(|| 0);
        let r = *counter % trunk_count;
        *counter += 1;
        return r;
    }
}