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VoIP capacity planning utility for sizing SIP trunks based on concurrent call model, codec selection, and quality profile — applicable to enterprise PBX, SBC, and carrier VoIP interconnects.
SIP trunk sizing converts busy-hour voice traffic into concurrent channels with Erlang B, then multiplies by codec bandwidth to estimate IP capacity. Use it before ordering ITSP trunks or provisioning an SBC so peak-hour blocking stays within your Grade of Service.
The sizer applies Erlang B traffic engineering to convert offered load (Erlangs) to channel count at a target blocking threshold, then computes aggregate RTP/UDP/IP bandwidth.
Built on 3GPP Technical Specifications, ensuring accuracy for production deployments and security audits.
Calculations specifically designed to identify protocol vulnerabilities and signaling misconfigurations.
Essential for SBC, carrier IP PBX, and ITSP interconnect planning. Accurate SIP trunk sizing prevents call blocking during peak hours and avoids overprovisioning costs.
Edge-V8 Isolate
Technical Audit
< 50ms (Global)
Secured by Auth0
Example: 1,200 busy-hour call attempts with 180 s average hold time → offered load = (1200 × 180) / 3600 = 60 Erlangs. At 1% blocking (B=0.01), Erlang B requires roughly 74 channels. With G.711 at ~87 kbps/call, plan ≈6.4 Mbps of priority voice bandwidth plus signaling and headroom.
Assumes Poisson arrivals and exponential holding times (Erlang B). Does not model queueing, IVR dwell, video sessions, or asymmetric codec negotiation. Always validate against measured BHCA from your CDR or SBC.
You need at least 100 SIP channels if every call is active at once. In practice, size from offered traffic in Erlangs (busy-hour call attempts × average hold time / 3600) and apply Erlang B at your target Grade of Service (often 0.01 or 1% blocking). For short calls with peaks, add headroom for CPS bursts and failover capacity.
A SIP trunk is the logical interconnect to your ITSP or carrier. Channels (or sessions) are the concurrent call capacity on that trunk. Marketing “trunk” packages often mean a capped channel count plus concurrent session limits and codecs.
G.711 (≈87 kbps per call with typical RTP/UDP/IP overhead at 20 ms) is the safest baseline for quality. G.729 and Opus reduce bandwidth but may require licensed or negotiated codecs on both ends. Size bandwidth after you know channel count × per-call IP rate.
Yes when blocked calls are cleared (no queue) — the usual SIP trunk / PRI model. Use Erlang C when calls wait in queue (contact centers). Mixing the models under- or over-provisions channels.
Channel math is independent of the SBC, but bandwidth and CPS must include media anchoring, transcoding, and TLS/SRTP overhead if enabled. Dimension the SBC session capacity separately from the trunk channel count.
Classic tool for determining circuit requirements with zero queuing.
Dimension P-CSCF, S-CSCF, and I-CSCF nodes for VoLTE/VoNR networks based on BHCA and concurrent sessions.
Estimate Opus codec nominal bitrates and total IP bandwidth including RTP/UDP/IP overhead.
Deep-Dive Into Carrier VoIP & IMS Architectures
Understand the core call flow mechanics. Learn SIP signaling optimization, IMS dimensioning standards, and media transport encryption directly on TelcoSec Academy.
SSO Link: Academy Node 0x93FF