Urgent - Retell Bug
# ❓・bug-reporta
at123456
12/18/2025, 11:13 AM@AI Evy @Colin - Integrations @ Retell AI @Bing - cofounder@Retell AI @Joshua - Solutions Director @Shah - support @ Retell AI
I've already contacted email support and received automated responses that don't address my infrastructure issue.
This is URGENT and blocking production:
Issue: SIP → LiveKit media bridge not processing inbound RTP from Twilio
Evidence:
- SIP signaling: ✅ Works
- SDP: ✅ Correct (RTP port 59174, PCMU)
- Media session: ❌ NEVER establishes (Twilio shows all metrics as "-")
- Agent cannot hear caller
Configuration verified:
- All transport methods (UDP/TCP/TLS) tried
- IP whitelisting complete
- Symmetric RTP enabled
- All credentials verified
- Localized URI tried
This is NOT a configuration issue. This is a Retell infrastructure problem.
Email support sent me generic tutorials. I need engineering escalation NOW.
Call SID: CAd8622da4da2d0d240a374e2ee5a8355a
Full PCAP and evidence ready.
Please escalate to SIP/telephony engineering team immediately.
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AI Evy
12/18/2025, 11:13 AMThanks for tagging me @at123456. I'll look through relevant sources and get back to you shortly.
AI Evy
12/18/2025, 11:13 AM### Answer
I understand this is a production-blocking issue. Unfortunately, I'm not able to escalate tickets or contact engineering teams directly.
For urgent engineering escalation, please:
- Visit [Contact Sales]() for direct business support
- Check [Retell AI GitHub]() for issue tracking
Relevant context from similar reports:
Other users have experienced RTP-related issues with Twilio SIP trunking. One suggestion from the community: check Twilio call logs with RTP debugging, ensure proper re-INVITE/SDP handling after transfers, and confirm codec compatibility (e.g., G.711) . [(1) ]()
Additionally, from a SIP/RTP standpoint, RTP servers typically "force a call drop" when the RTP stream is gone for over 60 seconds—LiveKit does implement this behavior. [(2) ]()
Since your SIP signaling works but media never establishes, and you've verified symmetric RTP and transport methods, this does appear to be beyond typical configuration issues. Your PCAP evidence should be valuable for the engineering team.
I'd recommend re-contacting support with explicit reference to the Call SID and attaching the PCAP, emphasizing this is an infrastructure-level RTP bridging issue rather than a configuration question.
AI Evy
12/18/2025, 11:13 AM
To learn more, see the sources I considered:
[[1] 🛑 SIP Call Transfer Issue – No Audio Received, Charged Despite RTP Timeout](https://discord.com/channels/1186424376300679209/1382138392862785597)
[[2] Livekit is not responding to BYE SIP messages](https://discord.com/channels/1186424376300679209/1432392693190889523)
AI Evy
12/18/2025, 11:13 AMFeel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
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Omar Ashraf | Community Lead
12/18/2025, 2:34 PMHello @at123456 , can you explain the problem you have instead of the chatpgt answers , because its not always right and the call id you shared is twilio call id , share a retell call id.
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at123456
12/18/2025, 2:47 PM@Omar Ashraf | Community Lead Hi, thank you for responding. I have this one-way audio issue with my agent, I can hear it but it cant hear me.
The call id:
call_2da986db40f3094f669a32de1e0.
o
Omar Ashraf | Community Lead
12/18/2025, 3:04 PMHello @at123456 , I just test it on my phone and it works fine
https://cdn.discordapp.com/attachments/1451170485227229224/1451228535690891378/image.png?ex=69456965&is=694417e5&hm=775d44519326a7cd3d73f908580293a9666bb4f68fc79fa89cfeb251124a21a0&
Omar Ashraf | Community Lead
12/18/2025, 3:04 PMThe issue could be on your phone
a
at123456
12/18/2025, 3:05 PMI am calling from GHL phone numbers, is that an issue ?
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Omar Ashraf | Community Lead
12/18/2025, 3:05 PMcould be, try a normal phone.