A single G.711 call uses about 87 kbps with headers. Fifty simultaneous calls use less than 5 Mbps. Voice never fails for lack of bandwidth — it fails because packets arrive late, out of order, or not at all, and because the signaling path and the media path disagree.
The numbers per call
| Codec | Payload | With RTP/UDP/IP | Notes |
|---|---|---|---|
| G.711 | 64 kbps | ~87 kbps | Best quality, most bandwidth |
| G.729 | 8 kbps | ~31 kbps | Compressed, more CPU |
| Opus | 6–40 kbps | varies | Modern, adaptive |
Add 20–30% for signaling, retransmits, and call setup bursts. Fifty G.711 calls fit comfortably in 5 Mbps.
What actually breaks calls
| Symptom | Likely cause | Where to look |
|---|---|---|
| One-way audio | Asymmetric routing, NAT/SIP ALG | Routing table, ALG settings |
| No audio at all | RTP ports blocked | Firewall policy for the media range |
| Robotic voice | Jitter, buffer starvation | Queue depth, congestion, Wi-Fi |
| Calls drop at 30 s | Session timer or state timeout | Firewall conntrack, SIP timers |
| Echo | Analog gateway, handset | Hardware, echo cancellation |
| Delayed dial tone | DNS or provisioning | Phone boot logs |
One-way audio is the classic: signaling works because it is TCP to the PBX, and media fails because the RTP path is asymmetric or NATed differently.
SIP ALG is usually the problem, not the solution
Many firewalls' SIP ALG rewrites headers incorrectly, producing registration failures and one-way audio. If your PBX is behind NAT, configure it with its external address and disable the ALG, rather than trusting the firewall to fix SIP.
Jitter and MOS
Voice quality is scored 1–5 (MOS). Practical targets:
| MOS | Experience |
|---|---|
| 4.3+ | Toll quality |
| 4.0 | Good |
| 3.6 | Noticeable but usable |
| 3.0 | Poor |
| < 3.0 | Unusable |
Jitter above 30 ms starts to exhaust the phone's buffer. Latency above 150 ms one-way makes conversation feel like a radio call.
Worked example: 100 Mbps circuit, 50 calls
| Item | Calculation |
|---|---|
| Media | 50 × 87 kbps = 4.35 Mbps |
| Signaling | ~0.2 Mbps |
| Headroom for burst | +30% |
| Total reserved | ~6 Mbps |
| QoS cap | 10 Mbps priority |
Cap the priority queue at 10 Mbps. That allows 50 calls plus headroom, and prevents a misbehaving device from using the entire priority class.
Call admission, simply
If the circuit can carry 50 calls with quality, the 51st should fail fast rather than degrade all 50. Some PBXs support admission control; if yours does not, monitor concurrent calls and alert before the limit.
What breaks in production
- NAT or ALG rewriting SDP, producing one-way audio.
- RTP blocked because only the signaling port was opened.
- No priority queue, so a backup saturates the link and every call degrades.
- Wi-Fi jitter, which no amount of wired QoS can fix.
- Session timers shorter than the firewall's state timeout, dropping calls at exactly 30 minutes.
Checklist
- Media ports are open end to end, both directions.
- The PBX knows its external address; the ALG is off.
- Voice is marked EF and capped at a calculated limit.
- Call capacity per circuit and per AP is known.
- MOS or jitter is measured during peak, not at idle.
- Session timers and firewall timeouts agree.
Next
Voice is the strictest case. Continue with Video and Screen Sharing, which is burstier.