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 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.
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 — rewriting SIP packet headers in ways that confuse your phone system, causing one-way audio, calls that drop immediately, or registration failures.
Log into your router admin panel and disable SIP ALG. On most routers it's under Advanced → NAT or Firewall settings. Reboot the router after. This alone resolves about 40% of VoIP issues I see.
If your team is on video calls or downloading large files while someone is on a VoIP call, voice packets compete for bandwidth with everything else. Voice is extremely sensitive — 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.
Speed isn't the issue most people assume. A VoIP call only uses about 85 kbps — almost any broadband connection handles 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. Anything above 30ms is problematic for business calling. Run the test here →
When a SIP device registers with your phone system, it punches a hole through your firewall. If your router closes that hole before the phone re-registers, inbound calls ring but produce silence — or drop immediately. This is tricky 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.
If your FreePBX offers G.722 (HD audio) but your SIP provider only supports G.711, call negotiation fails silently — resulting in one-way audio or an immediate hangup. Common after a SIP trunk migration where old codec settings get carried over.
Connectivity → Trunks → [your trunk] → SIP Settings → Codecs
Make sure your codec list matches what your SIP provider supports. G.711u (PCMU) is the universal fallback — put it first in the list if you're having issues.
That usually means the issue is further upstream — your ISP, a misconfigured firewall rule, or a SIP trunk registration problem. I've diagnosed and fixed these setups for businesses across Southern California. Send me your symptoms and I'll point you in the right direction.
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