You switched to VoIP to save money — not to spend every other call explaining to customers why they can't hear you. Dropped calls and one-way audio are the #1 complaints I hear from San Diego businesses after moving to a cloud phone system. The good news: almost every case comes down to one of five causes, and all of them are fixable.
01 SIP ALG Is Enabled on Your Router
SIP ALG (Application Layer Gateway) is a "feature" built into most consumer and small-business routers that tries to help with VoIP traffic. In practice, it almost always breaks it. It rewrites SIP packet headers in ways that confuse your phone system — resulting in one-way audio, calls that connect then immediately drop, or registration failures.
Log into your router admin panel and disable SIP ALG. On most routers it's under Advanced → NAT or Firewall settings. After disabling, reboot the router and test calls. This alone resolves about 40% of VoIP issues I see.
02 No QoS (Quality of Service) Configured
If your team is on video calls, streaming, or downloading large files while someone is on a VoIP call, voice packets are competing for bandwidth with everything else. Voice is extremely sensitive to delays — even 150ms of extra latency is noticeable to both parties.
UDP port 5060 → SIP signaling (highest priority) UDP 10000–20000 → RTP audio stream (high priority)
Enable QoS on your router and prioritize UDP port 5060 and your RTP port range. This tells the router to always let voice packets through first, regardless of other traffic on the network.
03 High Jitter on Your Internet Connection
Speed isn't the issue most people assume. A VoIP call only uses about 85 kbps per line — almost any broadband connection can handle that. The real culprits are latency (delay) and jitter (inconsistent delay). High jitter causes choppy, robotic audio and dropped packets mid-conversation.
Run a proper VoIP readiness test — not just a speed test. You need your jitter score specifically. Anything above 30ms is problematic for business calling. Run the test here →
04 NAT Timeout Too Short
When a SIP device registers with your phone system, it punches a hole through your firewall/NAT. If your router closes that hole before the phone re-registers, inbound calls ring on your end but produce silence — or drop immediately after answering. This is one of the trickier ones to diagnose because it only happens intermittently.
Set your router's UDP timeout to at least 180 seconds. Also set your SIP trunk registration interval to 60 seconds or less on your PBX so it keeps the NAT hole alive with regular re-registration.
05 Codec Mismatch Between PBX and SIP Trunk
If your FreePBX is configured to offer G.722 (HD audio) but your SIP provider only supports G.711, the call negotiation fails silently — resulting in one-way audio or an immediate hangup after the call connects. This is especially common after a SIP trunk migration where the old provider's codec settings get carried over.
Connectivity → Trunks → [your trunk] → SIP Settings → Codecs
Make sure the codec list in your trunk settings matches what your SIP provider supports. G.711u (PCMU) is the universal fallback that works with every provider. Put it first in the list if you're having issues.
// Still dropping calls after all five?
That usually means the issue is further upstream — your ISP, a misconfigured firewall rule, or a SIP trunk registration problem that only shows up under load. I've diagnosed and fixed these setups for businesses across San Diego. Send me your symptoms and I'll point you in the right direction.
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