Custom SIP / Bring-Your-Own-Dialer — UnlimCall Trunk Setup

Who this is for

Any team running outbound at volume on a platform not covered by a named integration guide. That includes:

• Proprietary predictive dialers built in-house or acquired — Asterisk-based, Dialogic, custom C++ engines
• SBC-fronted architectures where a session border controller sits between your dialer and the carrier
• Multi-tenant platforms where a single trunk feeds many customer sub-accounts
• VoIP resellers who want to offer flat-rate outbound to their own customers
• Developers building a new dialer who need a clean carrier integration from day one

If your platform originates SIP INVITE messages, this page has everything you need to connect.

Supported codecs

UnlimCall accepts and originates the following codecs in order of preference:

G.711 is the recommended default. Offer PCMU first in your SDP, PCMA second. Do not offer video codecs in the SDP — the trunk will reject the offer if no audio codec is negotiated.

Authentication methods

Two authentication modes are supported. Choose based on your network architecture:

IP-based authentication (recommended for fixed-IP deployments)

Whitelist your dialer's public-facing IP address (or CIDR range if you run multiple origination nodes) during onboarding. The trunk accepts INVITE without a SIP Authorization header from whitelisted sources. No registration keepalive required. This is the simplest and most operationally stable method for high-volume dialers where registration churn would create unnecessary load.

Registration-based authentication (for dynamic IP or cloud deployments)

Register via SIP REGISTER using the credentials provided at onboarding (username, password, SIP domain). Use a registration interval of 60–120 seconds. Send OPTIONS keepalives every 30 seconds if your dialer supports it — this allows the trunk to detect path failures before a call attempt. Registration-based auth is well-suited to dialers that scale horizontally across ephemeral cloud instances.

Transport and security

• UDP/5060: Default. No overhead. Suitable for internal-network or SBC-fronted deployments where the SBC terminates the external connection.
• TCP/5060: Supported. Recommended when UDP packet loss or fragmentation is a concern (large SDP bodies, heavy call volume).
• TLS/5061 + SRTP: Supported for encrypted media. Required for certain regulatory environments. Provide your dialer's TLS certificate or use IP-based trust with TLS transport. SRTP key exchange via SDES.

How to connect a custom SIP dialer to UnlimCall

1. Complete onboarding. UnlimCall provisions your local caller ID number(s) — one per market you activate — and provides your SIP domain, authentication credentials or IP whitelist, and the SIP trunk host. No shared DID pool; numbers are allocated for your account.

1. Configure your trunk peer. Point your dialer's outbound SIP peer or trunk definition at the UnlimCall SIP domain. Set the destination port (5060 UDP, 5060 TCP, or 5061 TLS). Set the contact/from-domain to your assigned SIP domain.

1. Set authentication. For IP auth: add your origination IP to your whitelist via the onboarding dashboard. For registration: configure your dialer's registrar with the supplied username, password, and realm.

1. Configure outbound caller ID. Set the From header or P-Asserted-Identity (PAI) header to the local number provisioned for the target market. The trunk will pass the number to the terminating carrier. For US/CA traffic, STIR/SHAKEN signing is applied at the trunk level using the provisioned number as the attested identity — no dialer-side signing required.

1. Set codec order in SDP. Ensure your dialer's SDP offer lists G.711 PCMU first (or PCMA for non-US markets). Strip unsupported or video codecs from the offer to avoid negotiation failures.

1. Handle failover. The trunk exposes a primary and secondary SIP endpoint. Configure your dialer to retry on the secondary endpoint on 5xx responses, timeouts, or transport errors. If your dialer supports DNS SRV, the trunk DNS record includes both endpoints in priority order.

1. Test with a low-volume call. Place a call, verify codec negotiation in your dialer's SIP trace, confirm the outbound caller ID on the receiving end, and check audio quality both ways. Typical post-dial delay from INVITE to 200 OK is under 1.5 seconds on US/CA routes.

What you get

Failover and resilience

UnlimCall operates multiple regional PoPs. Your trunk configuration will include a primary and secondary SIP host. For dialers that support DNS SRV resolution, configure your trunk domain using the supplied SRV record and let the DNS layer handle failover automatically. For dialers that use explicit host entries, configure the primary host first and the secondary as a fallback with a lower priority weight.

RTP media is anchored to the nearest PoP. Edge-to-origin audio latency is under 50ms for covered markets. If your dialer supports RTCP statistics collection, monitor round-trip-time and packet loss per leg — typical values on a healthy call are RTT <80ms, packet loss <0.5%.