Honestly, bridging SIP trunks with WebRTC has been a nightmare

So we've been building this voice AI agent that needs to work with regular phone calls, right? I thought the hard part would be making the AI sound natural. Wrong.

The actual headache? Getting SIP trunks to play nice with WebRTC.

Let me vent a bit and maybe save someone else the pain.

The basic problem

Your AI speaks WebRTC (modern browser stuff). Phone networks speak SIP (old telecom protocol from like the 90s). You need both because:

WebRTC = how you handle audio in web apps
SIP trunks = how you get actual phone numbers and connect to regular phones

Someone has to translate between them. That someone is you.

What's been killing us

Codec mismatches SIP trunk sends G.711. WebRTC wants Opus. Now you're transcoding in real-time which adds 20-50ms latency AND eats CPU like crazy. We were hitting 80% CPU per call before we figured out better codec negotiation.

NAT traversal issues One-way audio. The absolute worst. You hear them, they don't hear you. Or neither of you hear anything. Works perfectly in dev, completely broken in production. Spent a whole week on this before realizing our firewall was blocking the RTP port range.

DTMF is a mess When someone presses a phone key, SIP and WebRTC handle it completely differently. SIP uses RFC 2833, WebRTC doesn't support it directly. Had to build detection on the media server side and relay it through a data channel.

Call quality monitoring You can't fix quality issues if you don't know they're happening. Started monitoring packet loss, jitter, and RTT every second. Found out 15% of our calls had one-way audio because of NAT issues we didn't even know about.

What actually worked

Got it down to about 1% failure rate now. Here's what helped:

TURN servers - forced relay mode. Costs more bandwidth but eliminates NAT problems
Smart codec selection - prioritize PCMU/PCMA (less transcoding = less latency)
Multiple trunk providers - when one has issues, auto-failover to backup
Real-time monitoring - catching issues before users complain
Pre-warmed connection pools - avoid setup delays during high traffic

The stupid stuff that bit us

Spent 2 weeks optimizing the wrong thing (LLM latency) when network routing was the real bottleneck
Didn't realize enterprise SBCs strip WebRTC headers until we tried deploying at a client
One carrier's NAT was rewriting IP addresses randomly. Took forever to diagnose.
DTMF detection was at 85% accuracy. Turns out we needed dual-method detection (in-band + RFC2833)

Still struggling with