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Integration · 2 min read

Supported codecs, ports, and IP authentication

Network requirements for connecting to sip.non-cli.site and which codecs we negotiate.

Codecs

  • g711 ulaw / alaw — recommended, best quality, ~80 kbps/leg
  • g729 — supported, ~30 kbps/leg, slight quality loss
  • opus — supported on CLI tier, negotiated when both sides offer it

No transcoding fees; we pass codecs end-to-end where the destination supports them.

Ports and protocols

  • SIP signaling: UDP/TCP 5060 (TLS 5061 on request)
  • RTP media: UDP 10000–20000
  • Outbound from your softswitch must reach sip.non-cli.site on the SIP port and the destination IPs on RTP

Authentication

Two modes — pick one per trunk:

  1. Username + password (default). Works from any IP, registration-based.
  2. IP ACL. Lock the trunk to one or more static IPs; skips registration. Faster failover, no password to leak. Request via support.

TLS / SRTP

Available on CLI tier and on request for non-CLI. Adds ~10% to the per-minute rate due to the extra processing on our SBCs.

Related in Integration

How to integrate a SIP trunk with Asterisk
Working pjsip.conf and extensions.conf for non-cli.site trunks, with non-CLI and CLI examples.
How to integrate a SIP trunk with FreeSWITCH
Gateway XML and dialplan for non-cli.site termination with optional CLI override.
How to set custom caller ID (and stay legal)
When CLI override is allowed, how to set it in your softswitch, and what STIR/SHAKEN does to unattested calls.

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