Work through it, tick as you go.
Signalling & call control
Voice APIs are event-driven: the platform calls you as the call progresses.
- Understand the call lifecycle events (initiated, ringing, answered, completed) your API emits.Your webhook handler is effectively the call-flow state machine — design it as one.
- Model call control as idempotent webhook handlers that can tolerate retries and out-of-order events.
- Decide between declarative call-flow documents and live API control per event, per use case.
- Handle SIP concepts you can't avoid: early media, DTMF, transfer and re-INVITE semantics.
- Plan number strategy (E.164), caller-ID presentation and per-market caller-ID rules.
Media, codecs & quality
- Choose codecs deliberately: G.711 for quality, G.729/Opus where bandwidth is constrained.Budget bandwidth including IP/UDP/RTP overhead, not just the codec bitrate.
- Set a sensible packetization interval (commonly 20 ms) balancing overhead against latency.
- Design for jitter, packet loss and latency: jitter buffers, and realistic MOS expectations.
- Plan DTMF handling (RFC 2833 / SIP INFO / in-band) consistently across the path.
- Decide recording and transcription needs early — they affect media routing and storage.
Webhooks & integration
- Verify webhook signatures and reject unsigned or stale events.An unauthenticated call-control webhook is a takeover risk.
- Respond to webhooks fast and do heavy work asynchronously to avoid blocking the call.
- Make handlers idempotent and deduplicate on event IDs, since retries are expected.
- Correlate call legs with a stable call/session ID across transfer and conference.
Capacity & scaling
- Size concurrent channels from CPS × average call duration, then add peak headroom.Concurrent calls ≈ calls-per-second × average duration in seconds.
- Provision trunk / channel capacity and confirm provider concurrency and CPS limits.
- Scale webhook handlers horizontally and keep them stateless behind a queue.
- Load-test at peak concurrency and confirm graceful degradation, not cliff-edge failure.
Security, reliability & observability
- Encrypt signalling (SIP over TLS) and media (SRTP) where offered; restrict by IP allow-list.
- Protect against toll fraud and Wangiri: rate-limit and restrict high-risk premium destinations.
- Design failover across regions/providers so a single outage doesn't drop all calls.
- Log per-call detail records (CDRs) and events, and retain audio only per your compliance rules.
- Instrument quality metrics (MOS, ASR, ACD) and alert on drops per route.
- RFC 3261 · SIP: Session Initiation Protocol
- RFC 3550 · RTP: A Transport Protocol for Real-Time Applications
- ITU-T · G.711 codec
Standards and regulator references cited in this guide. BroadNet is a GSMA Associate Member and MEF member operating carrier-grade messaging since 2003.