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Moving from traditional telephone lines to SIP trunking reduces cost and increases flexibility, but the migration needs preparation. First-time adopters who skip readiness checks often discover problems during cutover: poor call quality, one-way audio, or security incidents. This checklist helps organizations plan the move before placing the first SIP call.
Pre-Deployment: Audit the Current Environment
- Inventory existing lines and numbers — document every PSTN line, ISDN channel, published number, and fax line
- Measure concurrent call volume — size SIP trunks to peak usage plus headroom, not total extension count
- Map call flows — record how inbound calls reach departments, after-hours routing, and emergency line behavior
- Identify analog dependencies — door phones, alarms, elevators, and fax machines may need gateways
- Review contracts — note termination clauses and number portability rules with current carriers
Network Readiness
SIP trunks ride the data network, so network quality determines voice quality. Verify bandwidth, latency, and jitter between your site and the SIP provider. Implement Quality of Service (QoS) policies that prioritize voice traffic over less time-sensitive data. Confirm firewall rules allow SIP signaling and RTP media in both directions, and consider dedicated network paths for voice where possible.
Security Before Go-Live
Internet-facing voice infrastructure attracts attackers. Before cutover, change default credentials, enforce strong SIP passwords, restrict administrative access to authorized networks, and enable TLS/SRTP encryption where supported. A Session Border Controller (SBC) adds a security boundary between your PBX and the public network, filtering malformed traffic and preventing toll fraud.
| Security Layer | Action |
|---|---|
| Authentication | Strong per-extension passwords; disable unused accounts |
| Network | Firewall rules limiting SIP access to known sources |
| Transport | TLS for signaling, SRTP for media encryption |
| Policy | Class-of-service restrictions on international and premium destinations |
| Monitoring | Alerts for unusual call volume or failed registration spikes |
Number Porting and Cutover
Number porting is the most common source of delay. Start the process early, request firm timelines from both old and new carriers, and plan a cutover window with rollback capability. Parallel running — where the new SIP trunks carry test traffic while old lines remain active — reduces risk before the final switch.
Post-Cutover Validation
After go-live, test every critical path: inbound DID routing, outbound dialing, emergency calls, fax machines, alarm lines, inter-site calling, and voicemail. Monitor call quality metrics for at least two weeks to catch issues that only appear under real traffic.
Redstone provides deployment planning support, gateway options for analog devices, and SBC solutions for secure SIP trunking boundaries.
Plan your SIP trunk migrationRedstone Systems, Inc. founded in Delaware, USA in December 2002, has been the ODM vendor for many well-known communications companies, serving the Southeast Asian market. In 2020, Redstone Systems will begin to return to the North America market with its self-developed brand.
Redstone has a complete product line of intelligent voice gateways, providing IP-PBXs, analog VoIP gateways (FXS/FXO), digital VoIP gateways (E1/T1), border appliances, and session boundary controllers (SBCs).
With advanced technology in digital signal processor (DSP), speech coding and speech processing, as well as efficient operational tools such as cloud remote management, auto provisioning, Redstone gateways are widely used in markets of enterprise communications, cloud communications, call centers, operators’ IMS/SIP trunks, bringing users friendly, efficient and reliable communication experience.
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