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    RingCentral SIP softphone telephony binding configuration.

    interface RingCentralTelephonyConfig {
        codec?: "OPUS/16000" | "OPUS/48000/2" | "PCMU/8000";
        costPerMinute?: number;
        ignoreTlsCertErrors?: boolean;
        inboundRunAsUserEmail?: string;
        maxCallSeconds?: number;
        maxConcurrentCalls?: number;
        outbound?: OutboundPolicySettings;
        sipAuthorizationId: string;
        sipDomain: string;
        sipOutboundProxy: string;
        sipPassword: string;
        sipUsername: string;
        transferTargets?: TransferTargetSettings[];
    }

    Hierarchy (View Summary)

    Index

    Properties

    codec?: "OPUS/16000" | "OPUS/48000/2" | "PCMU/8000"

    Codec to negotiate. Defaults to OPUS/16000.

    costPerMinute?: number

    Estimated carrier cost rate per minute in reporting currency (e.g. 0.015 for 1.5 cents/min). Used to compute Interaction.CostEstimate at call end. Defaults to 0.015 if unset.

    ignoreTlsCertErrors?: boolean

    Skip TLS cert validation (sandbox/test only — never in production).

    inboundRunAsUserEmail?: string

    Email of the user INBOUND calls run as. Required for inbound calls: if unset, unknown, inactive, or the system user, the call is rejected (there is deliberately no fallback to a privileged user). Point it at a dedicated least-privilege account.

    maxCallSeconds?: number

    Maximum length of one phone call in seconds (default 1800). The session is stopped at the cap.

    maxConcurrentCalls?: number

    Most phone calls (every carrier together, both directions) the server will carry at once (default 25). Keep this at or below the realtime model plan's concurrent-session limit: past the cap an inbound caller hears a polite "all agents are busy" and an outbound call is refused, instead of every call degrading.

    Outbound destination policy and per-user rate limit applied to every Place*Call mutation.

    sipAuthorizationId: string

    SIP authorization id (the device's RingCentral authorization id).

    sipDomain: string

    SIP domain (e.g. sip.ringcentral.com).

    sipOutboundProxy: string

    SIP outbound proxy (host:port, e.g. sip10.ringcentral.com:5096).

    sipPassword: string

    SIP auth password (resolved upstream — never inlined).

    sipUsername: string

    SIP auth username (the device's phone number / extension).

    transferTargets?: TransferTargetSettings[]

    The places the agent may transfer a live call to, by name. Empty or absent means the agent cannot transfer at all, even on a carrier that supports it: the agent never names a free-form number.