Direct Tier-1 carrier interconnects, instant elasticity, and automated multi-region failover. Connect your IP-PBX, contact center, or custom voice application with crystal-clear Opus audio quality.
Eliminate rigid on-premise PRI lines and expensive contract lock-ins with modern, elastic SIP trunking.
Pay only for the channels you actually use with automatic bursting. Scale from 10 to 10,000 concurrent calls instantly during peak traffic spikes.
Multi-homed IP connectivity with automated sub-second route rerouting. If a primary data center degrades, your calls instantly route to secondary nodes.
Ensure complete eavesdropping protection. Signaling is encrypted via TLS 1.3, and all RTP audio streams are protected with SRTP.
Transfer thousands of existing enterprise phone numbers seamlessly without a minute of downtime, backed by our dedicated LNP provisioning team.
Inspect jitter, packet loss, MOS scores, and call completion rates on live dashboard charts. Diagnose network issues before users ever complain.
Pre-certified and interoperable with Cisco CallManager, Asterisk, FreePBX, 3CX, Microsoft Teams Direct Routing, and Avaya Aura.
Provision test credentials in minutes with complimentary inbound and outbound calling credits.
SIP trunking is usually bought by someone who already has a PBX and wants to replace the lines feeding it, not the PBX itself. The questions below are the ones that actually determine whether a migration goes smoothly.
A PRI circuit is a physical line from the carrier into your building, bundling a fixed number of channels — typically 23 voice channels in North America — whether or not you use all of them. A SIP trunk replaces that physical circuit with a connection over your internet link, and instead of buying a fixed bundle of wires, you buy a number of concurrent channels that can be scaled up or down as a configuration change.
The underlying service your PBX dials through looks almost identical from the PBX's perspective — it is still SIP signaling in, SIP signaling out. The difference is in what's behind it: no physical circuit to provision, no on-site hardware tied to a fixed channel count.
Size trunks to concurrent calls, not headcount. A 50-person office rarely has 50 people on the phone at once — what matters is your actual peak: how many calls are active simultaneously during your busiest hour, including any overflow for campaigns, seasonal spikes or an outage at another site routing traffic to you.
Bandwidth sizing follows the same logic. On the Opus codec, each concurrent call uses roughly 50 kbps, so 20 concurrent calls need on the order of 1 Mbps of dedicated, low-latency capacity — not the advertised speed of your internet plan, which is shared with everything else on the network. Undersizing either the channel count or the bandwidth produces the same symptom: calls that fail to connect or that connect with poor audio during busy periods.
In most cases, yes. Salamtalk's SIP trunks interoperate with Cisco CallManager, Asterisk, FreePBX, 3CX, Microsoft Teams Direct Routing and Avaya Aura, covering the large majority of business PBX deployments in active use.
What actually varies between PBX platforms is configuration detail — codec negotiation, registration behavior, how the PBX expects authentication to work — not whether trunking works at all. Confirm your specific PBX version against our setup guides, or ask our team to validate it, before a cutover date is locked in.
Trunks route across geo-redundant infrastructure, so if one region degrades, calls shift to another without the PBX needing to be reconfigured mid-incident. This runs under the same 99.9% uptime SLA that covers the rest of the platform.
That redundancy covers our infrastructure, not your building's internet connection or your PBX hardware. A SIP trunk cannot route calls to a PBX that has no path to reach the internet, so if continuity during a local outage matters, the mitigation (a secondary circuit, or a documented forwarding path) has to live on your side of the connection too.
The honest comparison is about cost structure, not a specific price gap — real PRI pricing varies too much by region and carrier contract to generalize responsibly. The structural differences are real and consistent, though:
| PRI | SIP trunking | |
|---|---|---|
| Unit sold | Fixed bundle of physical channels | Concurrent channels, adjustable |
| Scaling | New circuit install for more capacity | Configuration change |
| Hardware | On-site circuit termination gear | None required beyond your existing PBX |
| Idle capacity | Paid for whether used or not | Sized to actual concurrent need |
The practical effect is that SIP trunking tends to avoid paying for unused channel capacity and avoids a hardware-driven delay when you need more channels. Whether that nets out cheaper for your business depends on your current PRI contract, which we won't guess at — get a quote against your actual channel count.
No. SIP trunking typically replaces the physical lines feeding an existing PBX, not the PBX itself. Salamtalk trunks interoperate with Cisco CallManager, Asterisk, FreePBX, 3CX, Microsoft Teams Direct Routing and Avaya Aura.
Size to your actual peak concurrent calls, not your headcount. Most businesses need far fewer channels than employees, since not everyone is on the phone simultaneously.
On the Opus codec, each concurrent call uses roughly 50 kbps. Size dedicated, low-latency bandwidth to your peak concurrent call count, not to your total internet plan speed, which is shared with other traffic.
The cost structures differ: PRI sells a fixed bundle of physical channels whether used or not, while SIP trunking sells adjustable concurrent channels. Whether that is cheaper depends on your current contract and actual usage, so get a quote against your real channel count rather than assuming a fixed savings figure.
SIP trunking depends on your connection to the internet, same as any cloud service. Our infrastructure fails over across regions on our end, but that does not restore connectivity at your building, so a secondary circuit or forwarding plan is worth having if continuity during local outages matters.