my sbc is behind NAT, what setup should i use on sbc. is nat traveral supported?

[tardy sedge]

AI-generated Answer for ivanx0929

Short answer

• NAT traversal: The SIP trunking docs do not describe any built‑in NAT traversal (e.g., STUN/TURN). Instead, they advise opening RTP ports and checking for NAT-related one‑way audio. Plan to handle NAT on your SBC/firewall. (elevenlabs.io)

Recommended SBC setup behind NAT (SIP trunking to ElevenLabs)

• SIP signaling transport: Use TCP or TLS; UDP signaling is not available. Open 5060/TCP (or 5061/TLS) and ensure your firewall allows the chosen transport. (elevenlabs.io)
• SIP destination (to ElevenLabs): Dial sip:rfc3261 URIs to sip.rtc.elevenlabs.io on TCP 5060 as documented. (elevenlabs.io)
• Where ElevenLabs sends calls to you: In the ElevenLabs phone number config, set the Address to a hostname or IP that is reachable from the public internet (your SBC’s public FQDN/IP). (elevenlabs.io)
• Authentication: Use digest auth if possible; ACL/IP‑allowlisting is also supported. (elevenlabs.io)
• Static IPs for allowlisting (signaling): If you require static source IPs, use the static endpoint(s) provided for Enterprise (sip-static.rtc.elevenlabs.io, and region-specific options). You must allowlist the entire /24 block. (elevenlabs.io)
• RTP/media through NAT: Allow bidirectional UDP for RTP (typically ports 10000–60000). One‑way/no audio is commonly due to NAT or blocked RTP. Do not restrict RTP to specific static IPs, as RTP may use dynamic IPs. (elevenlabs.io)
• If you’re behind NAT, disable SIP ALG on the router/firewall; it can break media/signaling and cause one‑way audio. (elevenlabs.io)
• Codecs: Ensure codec compatibility (G.711 µ/ A‑law 8 kHz or G.722 16 kHz supported on ElevenLabs’ SIP side). (elevenlabs.io)

Notes

• ElevenLabs runs multiple SIP servers behind a load balancer; subsequent in-dialog requests should target the Contact URI you received (helpful if you see 481 on BYE). (elevenlabs.io)

References

• SIP trunking docs (requirements, transports, RTP ports, dialing, auth, troubleshooting, codecs): https://elevenlabs.io/docs/eleven-agents/phone-numbers/sip-trunking (elevenlabs.io)
• Static IP signaling endpoints and /24 allowlisting (Enterprise): https://elevenlabs.io/docs/eleven-agents/phone-numbers/sip-trunking (elevenlabs.io)
• Telnyx integration troubleshooting (disable SIP ALG, open RTP UDP, codec compatibility): https://elevenlabs.io/fr/agents/integrations/telnyx (elevenlabs.io)

If you need specifics for your carrier/SBC model, share your signaling transport choice (TCP/TLS), public FQDN/IP, and firewall policy and I can map them to the exact fields

[royal talon]

i use tcp signaling. I have successfully configured a SIP TCP 5060 connection. My firewall and Cisco CUBE logs show that the TCP handshake is ESTABLISHED, but the SIP INVITE is being silently discarded or the connection is closed immediately without a SIP response (no 401 Unauthorized or 100 Trying).