Fill in a short form — get production-shape IOS-XE CUBE config for your SIP trunk: voice class codec, sip-profile, tenant, and inbound / outbound dial-peers.
The SIP Dial-Peer Generator produces a ready-to-paste IOS-XE CUBE configuration for a typical enterprise SIP trunk: inbound and outbound dial-peers, a shared voice class codec, a voice class sip-profiles block for common header manipulations, and an optional voice class tenant for 17.x-family images. You pick the destination IP, codec list, DTMF method, SRTP, T.38, and a dial string prefix — the tool writes the config with comments you can keep in place for your change record. Use it to scaffold a net-new trunk, to onboard a second carrier, or as a sanity check when inheriting a messy existing config.
Fill in the form and click Generate config — the IOS-XE config appears here.
Apply the blocks in the order they appear: voice service voip, voice class codec, voice class sip-profile, voice class tenant, then the dial-peer stanzas. Each block refers to the ones above it by number.
Cisco offers codecs in order of preference. Put the cheap codec (G.729) first only if your carrier genuinely supports it on the inbound side — otherwise you end up in codec negotiation loops.
The inbound dial-peer exists so the router knows how to classify an incoming INVITE. Without it, CUBE falls back to dial-peer 0 and you'll see mysterious one-way audio. The generator always emits both.
CUBE configs fall into a handful of patterns. The one this tool generates is the SIP-trunk-as-tenant pattern — it scales from one carrier to N carriers with only incremental change. The shared voice class codec means every future dial-peer references codec preference by ID rather than inline, so when a carrier's contract changes you touch one line. The shared voice class sip-profiles holds the header manipulations you will eventually need — the tool seeds the three most common ones (strip From host to hide your public IP, force PAI from calling-party, downgrade unwanted SDPs).
The voice class tenant construct was introduced in IOS-XE 17.1 and makes multi-carrier life much easier. It lets you scope sip-server, session target, and credentials per trunk instead of globally. If you're on 16.x, comment out the tenant block — the rest of the config still works if you move the sip-server and session target statements up to sip-ua / directly into the dial-peer.
The inbound and outbound dial-peers are deliberately simple. Outbound matches on your dial prefix and session target ipv4: points at the carrier; inbound matches on the carrier's destination IP with incoming uri via. You will almost certainly layer translation-profiles, class-of-restriction (COR), and CAC on top — this config gives you the scaffold that works before those policies turn it into a wall of text.
If you enable SRTP, the generator adds srtp under the dial-peer and the matching srtp crypto 1 reference. Your carrier must support SRTP (most ITSPs do, some mid-market carriers don't); if they don't, you'll see 488 Not Acceptable Here on call setup. T.38 similarly requires carrier support — the generator emits the fax protocol config but if your carrier is G.711 pass-through only, comment out the T.38 lines.
Yes. The generated config runs on IOS-XE 16.9+ CUBE on ISR 4000, Cat 8000V, Cat 8200L/8300, and CSR1000v. IOS 15.x CUBE on older ISR G2 is supported for the core dial-peer constructs but the voice class tenant block should be omitted.
RFC 2833 / rtp-nte is the carrier interoperability default — it works with virtually every ITSP and with CUCM. sip-notify is reserved for carriers that explicitly require it, and sip-kpml is Cisco-phone-specific and not portable.
voice class codec lets you keep codec preferences centralized — one line referenced from every dial-peer. When the carrier tomorrow says "also allow G.729", you edit the voice class once instead of touching every dial-peer in the config.
Tenants (IOS-XE 17.x+) isolate SIP behavior per trunk — useful when one CUBE fronts several carriers. The tool includes a tenant scaffold; comment it out if you're on IOS-XE 16.x.
It's a strong starting point for a standard SIP trunk. Every environment has local policy (translation rules, calling-party number normalization, call admission control, high-availability) that the tool doesn't know about — review with your voice engineer before paste.
Cat 8300, single ITSP, G.711 + G.729, SRTP on.
show dial-peer voice summary.Two ITSPs, active/standby. Primary dial-peer preference 1, secondary preference 2.
preference on the outbound dial-peers to prefer carrier 1.SIP trunk between CUCM and the CUBE, G.711 only, T.38 for faxing.
fax protocol t38 version 0 on both sides.The Network & UC Health Assessment is a written, independent review of your actual environment — not vendor marketing. Includes a SIP trunk audit covering codec negotiation, DTMF, SRTP, fax handling, CAC, and redundancy.