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URGENT: SIP → LiveKit Media Bridge IssueI have a c...
# ❓・ai-evy-support
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@AI Evy URGENT: SIP → LiveKit Media Bridge Issue I have a confirmed infrastructure issue with custom telephony integration. PROBLEM: - SIP signaling works ✅ - SDP correct (RTP port 59174) ✅ - But RTP media never establishes - Twilio shows all metrics as "-" - Agent cannot hear caller EVIDENCE: - All configuration verified correct (matches docs exactly) - Tested from multiple sources (same issue) - Call SID: CAd8622da4da2d0d240a374e2ee5a8355a - Your SDP shows LiveKit backend - RTP from Twilio not reaching LiveKit session This is NOT a configuration issue - this is a SIP → LiveKit media bridge failure. Can someone from engineering investigate? This is blocking production. I have PCAP files, screenshots, and full evidence ready.
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Thanks for tagging me @at123456. I'll look through relevant sources and get back to you shortly.
### Answer --- Based on similar cases in the community, here are the most common causes for SIP signaling working but RTP media not establishing [(1) ]() : Common Causes: - NAT/firewall blocking RTP (open the IP/ports in LiveKit's SDP) - SIP ALG interfering (disable it) - Codec mismatch (use PCMU/PCMA/Opus) - Wrong SIP URI or region endpoint Recommended Checks: 1. Verify the 200 OK SDP and PCAP to confirm RTP is actually being sent both ways 2. Ensure Retell's IP block is whitelisted: 
18.98.16.120/30
(All regions) , 
143.223.88.0/21
(US only) , 
161.115.160.0/19
(US only) [(2) ]() SIP Server Info: - Retell SIP server URI: 
sip:sip.retellai.com
- Supported transport: TCP (Recommended) , UDP, TLS - Media encryption: SRTP (Transport should be set to TLS to use SRTP) For Dial-to-SIP Method: If you're using the Register Phone Call API approach, the SIP URI format should be: 
sip:{call_id}@sip.retellai.com
Since you have PCAP files ready and this appears to be blocking production, I'd recommend opening a support ticket with your evidence: - Visit [Contact Sales]() - Visit [) The engineering team can investigate the specific call_id and LiveKit session from their side.
​ To learn more, see the sources I considered: [[1] SIP call using call_id reaches LiveKit but no audio / agent not triggered](https://discord.com/channels/1186424376300679209/1389990836992872471) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
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@User Thanks for the response. I've already checked most of those: ✅ Codec: PCMU (verified correct) ✅ SIP URI: sip:sip.retellai.com (verified correct) ✅ Tested from multiple sources/networks (same issue) ✅ PCAP shows SIP works, but RTP never establishes NEW CHECKS I'LL TRY: - Whitelist Retell IPs in Twilio (18.98.16.120/30, etc.) - Try TLS transport instead of UDP BUT: The issue is that Twilio shows ALL metrics as "-" (no media session established at all). This suggests RTP is never being anchored, not just blocked. I'll open a support ticket with: - Call SID: CAd8622da4da2d0d240a374e2ee5a8355a - Full PCAP files - Configuration screenshots - Detailed evidence Can you confirm the ticket will go to engineering team? This is blocking production. Thanks!
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Appreciate the feedback! Feel free to ask me again with additional guidance or context.
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@AI Evy Thanks for the response. I've already checked most of those: ✅ Codec: PCMU (verified correct) ✅ SIP URI: sip:sip.retellai.com (verified correct) ✅ Tested from multiple sources/networks (same issue) ✅ PCAP shows SIP works, but RTP never establishes NEW CHECKS I'LL TRY: Whitelist Retell IPs in Twilio (18.98.16.120/30, etc.) Try TLS transport instead of UDP BUT: The issue is that Twilio shows ALL metrics as "-" (no media session established at all). This suggests RTP is never being anchored, not just blocked. I'll open a support ticket with: Call SID: CAd8622da4da2d0d240a374e2ee5a8355a Full PCAP files Configuration screenshots Detailed evidence Can you confirm the ticket will go to engineering team? This is blocking production. Thanks!
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Thanks for the follow-up, @at123456. I'll get back to you soon.
### Answer --- ### Additional Configuration Checks Since SIP signaling works but RTP never establishes, here are a few more things to verify: Transport Configuration: For different transport methods, the SIP server URL format changes [(1) ]() : - TCP: 
sip:sip.retellai.com;transport=tcp
- UDP: 
sip:sip.retellai.com;transport=udp
- TLS: 
sip:sip.retellai.com;transport=tls
For SRTP (media encryption) : Transport should be set to TLS to use SRTP Twilio-Specific Checks: - Check your Termination SIP URI for any spaces and remove them [(2) ]() - Use a localized termination URI near your region - Verify your username and credentials are correct (the username is different from the friendly name) Origination Setup: Ensure Retell's SIP server address is set as the origination SIP URI: 
sip:sip.retellai.com
--- ### Regarding Support Ticket I don't have information about the internal routing of support tickets to engineering teams. For production-blocking issues with detailed evidence like yours, I'd recommend: - Visit [Contact Sales]() - Visit [) You may also want to check Twilio's logs directly for any errors, as the documentation suggests: "Please check the logs in your telephony provider, and perhaps open a ticket with them." [(1) ]()
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Twilio](https://docs.retellai.com/deploy/twilio) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
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@AI Evy I have exhausted ALL configuration options for my SIP trunk integration. The issue persists and is blocking production. ## Problem: - SIP signaling: ✅ Works (200 OK, auth succeeds) - SDP: ✅ Correct (RTP port 59174, PCMU) - Media session: ❌ NEVER establishes (Twilio shows all metrics as "-") - Result: Agent cannot hear caller (one-way audio) ## What I've Verified: ✅ Symmetric RTP: Enabled ✅ IP ACL: Added (all 3 Retell IP ranges) ✅ Transport: Tried UDP/TCP/TLS (none work) ✅ Credentials: Verified (username/password match) ✅ URIs: Correct (sip:sip.retellai.com, retell-keey1.pstn.twilio.com) ## Evidence: - Twilio call logs: All media metrics show "-" (no media session) - PCAP: RTP packets sent by Twilio, never received by Retell - Symptom: Caller hears agent, agent cannot hear caller ## Root Cause: This is a Retell infrastructure issue. Your SIP → LiveKit media bridge is NOT processing inbound RTP from Twilio. This is NOT a Twilio configuration issue. ## Request: Please escalate to engineering immediately. This requires investigation of: 1. SIP → LiveKit media bridge 2. RTP packet processing on Retell edge 3. LiveKit session binding for SIP trunks This is blocking production deployment. I have full evidence ready. Thank you
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Thanks for the follow-up, @at123456. I'll get back to you soon.
llai.com/app-partner/jambonz) if you continue having connectivity issues—they provide a system that already connects with Retell and supports static IP addresses. Would you like a general answer about SIP/RTP troubleshooting approaches based on my knowledge?
### Answer --- I understand you've exhausted the standard configuration options. Based on your symptoms (SIP signaling works, but RTP media never establishes with Twilio showing all metrics as "-") , here are a few final checks from the documentation: Verify Origination Setup in Twilio: Ensure you've set up origination (for inbound) with Retell's SIP server address as the origination SIP URI: 
sip:sip.retellai.com
[(1) ]() Localized Termination URI: You might want to use a localized termination URI that's near your region. You can expand and view your localized URIs in the Twilio console. IP Blocks for Traffic: Confirm these are whitelisted correctly: [(2) ]() - 
18.98.16.120/30
(All regions) - 
143.223.88.0/21
(United States only) - 
161.115.160.0/19
(United States only) Note: Some non-US traffic might go from IPs not listed above, but will be migrated to the first IP block by 12/03/2025. From Community Reports: Similar issues have been reported where SIP signaling works but no audio establishes. Common causes identified include NAT/firewall blocking RTP, SIP ALG interfering, and codec mismatch. [(3) ]() --- For Production-Blocking Issues: The documentation states: "Please check the logs in your telephony provider, and perhaps open a ticket with them." [(2) ]() I don't have information about internal escalation paths to engineering teams. For urgent production issues with detailed evidence like yours, please reach out directly: - Visit [Contact Sales]() - Visit [) You may also want to consider Retell's partner [Jambonz](
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] SIP call using call_id reaches LiveKit but no audio / agent not triggered](https://discord.com/channels/1186424376300679209/1389990836992872471) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
a
@AI Evy I've tried the localized URI. Still doesn't work. I've now exhausted ALL configuration options: - All transport methods (UDP/TCP/TLS) - IP whitelisting (all 3 ranges) - Symmetric RTP - Localized termination URI (retell-keey1.us1.pstn.twilio.com) - All credentials verified Issue persists: Twilio shows no media session (all metrics "-"). Agent still cannot hear caller. This is 100% a Retell infrastructure issue with SIP → LiveKit media bridge. I've contacted Sales and opened a GitHub issue. Please escalate to engineering immediately. This is blocking production. Full evidence ready.
a
Thanks for the follow-up, @at123456. I'll get back to you soon.
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
### Answer I understand you've exhausted all standard configuration options and this is blocking production. Based on the available documentation, I've provided all the troubleshooting steps that exist: What the documentation covers: - SIP server URI: 
sip:sip.retellai.com
[(1) ]() - IP blocks: 
18.98.16.120/30
(All regions) , 
143.223.88.0/21
(US only) , 
161.115.160.0/19
(US only) - Transport methods: TCP, UDP, TLS - SRTP requires TLS transport For issues not resolved by configuration: The documentation states: "Please check the logs in your telephony provider, and perhaps open a ticket with them." Alternative option mentioned: If you're having trouble with the connection, you can check out Retell's partner [Jambonz]() , which provides a system that already connects with Retell and supports static IP addresses. Dedicated Slack support is available for Retell users through Jambonz. I don't have information about internal engineering escalation paths or the ability to escalate tickets. Since you've already contacted Sales and opened a GitHub issue, those are the appropriate channels for production-blocking infrastructure issues.
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
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