@Colin - Integrations @ Retell AI URGENT: Technical Bug - RTP Port 5060 in SDP Breaking Twilio SIP Trunk Your SIP stack is generating invalid SDP that breaks Twilio SIP trunking. The Problem: - Retell's 200 OK SDP shows:

m=audio 5060 RTP/AVP 0 8 101

- Port 5060 is the SIP signaling port, NOT RTP media port - This violates RFC 3264 (SDP) and breaks Twilio's SIP trunking - Result: Zero RTP packets transmitted → One-way audio Technical Evidence: - Call SID: CA74deb543b2c2923e824b929db1db9692 - Phone Number: +12136931521 - PCAP analysis shows 0 RTP packets (filtered for "rtp") - SIP signaling works (INVITE, 200 OK, ACK, BYE all present) - But no RTP packets ever transmitted Expected SDP: m=audio 10000 RTP/AVP 0 8 101 ← Should be 10000+, not 5060 What I've Verified: - ✅ Twilio SIP Trunk: Correctly configured - ✅ Origination URI: sip:sip.retellai.com (correct) - ✅ Termination URI: Configured with authentication (correct) - ✅ Retell phone number: Custom telephony configured (correct) - ✅ No media port settings available in Retell UI Request: 1. Please escalate this to a SIP engineer 2. This is a backend bug in Retell's SDP generation 3. Need confirmation that Retell's SIP stack will be fixed to use valid RTP ports (10000+) 4. Timeline for fix I can provide: - Full PCAP file - Detailed Wireshark analysis - Any additional technical details needed This is blocking production deployment. Please have a human engineer review this issue. Thank you!