Determine how many SIP trunk channels your business needs using the Erlang B formula, then calculate exact bandwidth requirements for your chosen codec.
Works with Asterisk, FreePBX, 3CX, and any SIP-compatible PBX
Using the Erlang B formula to determine required SIP trunk channels
Calculate the internet bandwidth required for your SIP trunk
Pre-filled from channel calculator above
Adds protocol header overhead for accurate real-world bandwidth
T.38 fax uses approximately 40 kbps per channel
IPComms has no channel limits - pay only for what you use. Unlike providers that sell fixed channel bundles, IPComms scales with your actual usage at just $0.01/minute for US and Canada calls.
The Erlang B formula is the industry-standard method for calculating the number of trunk lines (channels) needed to handle a given amount of telephone traffic with a specified blocking probability. Named after Danish mathematician A.K. Erlang, this formula has been used in telecommunications engineering since the early 20th century.
The formula calculates the probability that a call will be blocked (rejected) given a specific number of channels and a known traffic load measured in Erlangs:
B(n, A) = (An / n!) / Σk=0..n (Ak / k!)
B = blocking probability (the chance a call is rejected)
n = number of channels (trunks)
A = traffic intensity in Erlangs
Traffic intensity in Erlangs is calculated by multiplying the number of calls per hour by the average call duration in hours. For example, 30 calls per hour with an average duration of 3 minutes gives: 30 x (3/60) = 1.5 Erlangs.
A small office with 25 employees makes an average of 30 calls per hour during the busy period, with each call lasting 3 minutes:
The Erlang B formula assumes that blocked calls are cleared (the caller does not retry immediately). This is the most common model for sizing SIP trunks because callers typically wait before retrying or call back later.
For more on capacity planning, read our guide on SIP Trunk Capacity Planning.
Each voice codec compresses audio differently, resulting in different bandwidth requirements. The total bandwidth includes the codec payload plus IP, UDP, and RTP headers (40 bytes per packet).
| Codec | Payload Only | With Overhead | Quality | Best For |
|---|---|---|---|---|
| G.711 ulaw/alaw | 64 kbps | 87.2 kbps | Excellent | Most SIP deployments, LAN |
| G.729 | 8 kbps | 31.2 kbps | Good | Low bandwidth, WAN links |
| G.722 HD | 64 kbps | 87.2 kbps | HD Quality | HD voice, internal calls |
| Opus | 6-510 kbps | ~40 kbps | Excellent | WebRTC, modern endpoints |
| T.38 (Fax) | Variable | ~40 kbps | N/A | Fax over IP |
Overhead calculation assumes 20ms packetization interval with 40 bytes of IP (20) + UDP (8) + RTP (12) headers per packet.
Note: Call center assumes high utilization (50% of agents on calls simultaneously). Enterprise assumes lower utilization (~10% peak). Actual costs depend on call patterns. For a deeper comparison, see our SIP Trunking vs. PRI guide.
Configure your network equipment to prioritize voice traffic using Differentiated Services Code Points:
SIP Signaling: DSCP 24 (CS3)
RTP Media: DSCP 46 (EF - Expedited Forwarding)
Most IP phones and softphones support adaptive jitter buffers. Recommended settings:
For a typical office with 50 employees, you'll need approximately 8-12 SIP trunk channels. This assumes average call volumes of 2-3 calls per employee per hour during peak times with an average call duration of 3 minutes. Using the Erlang B formula with a 1% blocking probability gives you the precise number. Use the calculator above with your specific call patterns for an accurate result. With IPComms, there are no channel limits - you pay only for minutes used.
Each SIP trunk channel requires between 31.2 kbps (G.729 codec) and 87.2 kbps (G.711 codec) of bandwidth in each direction, including IP/UDP/RTP overhead. For 10 concurrent calls using G.711, you'd need approximately 1.74 Mbps of dedicated bandwidth. We recommend adding a 20% QoS buffer and ensuring your internet connection can handle voice traffic with low latency (under 150ms) and minimal jitter (under 30ms).
No, IPComms does not impose channel limits on SIP trunks. Unlike traditional providers that sell fixed channel bundles, IPComms uses a pay-per-minute model at $0.01/minute for US and Canada calls. Your concurrent call capacity is limited only by your PBX hardware and available bandwidth, not by arbitrary provider restrictions. This means you never pay for unused channels and can handle unexpected call spikes without service interruption.
G.711 (ulaw for North America, alaw for international) is the most widely used codec for SIP trunking because it provides excellent voice quality with no compression artifacts. If bandwidth is limited, G.729 uses only about one-third the bandwidth but requires licensing on some platforms. For HD voice quality, G.722 provides wideband audio at the same bandwidth as G.711. Opus is a modern codec offering excellent quality at low bandwidth (around 40 kbps) but requires both endpoints to support it. IPComms supports all major codecs.
No channel limits, no contracts. Pay-as-you-go pricing starting at $0.01/min for US and Canada.