Provide inter-office exchange, outgoing long-distance and international calls, incoming DIDs for enterprise customers and call centers
What is SIP trunking and how does it work?
SIP is a popular VoIP protocol and de-facto standard in today’s technological world of communications. SIP trunking allows IP PBX systems to connect various locations of a company’s offices by using VoIP (as opposed to sending calls via a traditional PSTN network and thus paying local telecom operators).
So, for example, a call travels over the Internet from an IP PBX in office A to the Internet telephony service provider (ITSP) VoIP Softswitch – where it is then routed to the IP PBX in office B. There are zero per-minute costs for these calls and the service is typically billed based on the maximum allowed number of connected calls (these are “trunks”).
Although standard SIP trunking usage allows companies to save costs on calls between extensions / employees – the real benefit comes from its ability to route outgoing long-distance and international calls via a VoIP at a reduced rate – or easily allocate phone numbers (local, toll-free or international) for inbound calls. In addition to connecting voice calls, - a frequently sought after value-added service is fax delivery via VoIP using a T.38 protocol, which ensures quality and best of all, saves costs.
What motivates enterprises to sign up for SIP Trunking?
Target customers are enterprises that have on-premises IP PBX, legacy PBX (connected to VoIP gateway) or Microsoft Lync. The three main business drivers are:
- Customer has multiple branch offices in various cities – so they want a toll bypass for calls between different branches;
- Customer wants to save money on outgoing long-distance and international calls; and
- Customer wants to have various incoming phone numbers (toll-free, in different cities or even different countries) and / or advanced features (such as call recording).
Architecture of a SIP trunking platform
For all that, you only need one single program: PortaSwitch. Its unified service delivery platform, includes the following components:
Session-Border Controller (SBC)
SBC provides a high-availability control point for all network communications, to keep the service continually accessible. It protects your network from denial-of-service (DoS) attacks (that can occur when either a malicious or misconfigured device attempts to flood the network with messages). It also limits call initiation ratio per end-point (IP PBX or SIP phone) – allowing you to properly allocate the call capacity to individual customers. This way a customer who initiates a sudden call burst (e.g. from an outbound call center campaign) does not affect other customers.
SIP Softswitch cluster
This highly scalable class 4 and 5 SIP Softswitch Cluster provides call control and media processing functions. It supports:
- Real-time call authorization and charging – thus enforcing the credit limit for long-distance and international calls;
- The control of simultaneously connected calls (active SIP trunks) with flexible options that allow you to apply limits to either all or only outgoing / incoming calls;
- Dynamic fraud monitoring and prevention – in case a customer's IP PBX equipment is hacked externally or abused by an employee. In that case, suspicious traffic is promptly detected and blocked;
- Dynamic call routing – this gives you the ability to select outgoing carriers according to a desired quality level at an optimized cost; and
- Voice and video calls, fax-over-IP (T.38) and SMSs (e.g. for marketing campaigns or customer alerts).
Integrated Billing (B/OSS)
Integrated Billing is used for configuring a customer’s service parameters such as methods of authorization, credit limits, “normal calling” profiles (in order to quickly detect any abnormal activity), number of purchased trunks and calls-per-second thresholds. There is a real-time data link between B/OSS and Softswitch, so any changes are instantly effective, no additional configuration is required.
This makes it easy to implement a variety of revenue-generating methods for SIP trunking billing, such as:
Charging basic recurring fees based on the number of trunks;
Per-minute charging or discounted bundles for long-distance and international calls;
Permitting "soft" and "hard" trunk limits, wherein all usage below the "soft" limit is free; cases wherein the number of active trunks goes rise above the "soft" limit (but stays below the "hard" onelimit) are allowed, but and in that case, the customer is charged on a per-incident basis. Once the "hard" limit is reached, and nno more calls can beare permitted to be established beyond the "hard" limit.
At the end of the billing period all the charges are combined into a consolidated invoice. Any applicable discounts or “committed usage” charges are applied; taxes (local and FCC) are assessed; then invoices are generated and distributed to customers automatically. On their self-care portal, customers can download a copy of their invoice, change their service configuration, browse their call history and also submit online payments via their credit card or an ACH bank debit.
Steps to become a SIP Trunking provider
To begin with, install PortaSwitch in a private cloud (on your own servers) or sign up for a PortaSwitch SaaS in a public cloud. Since reliability is very important in support of the business telephony services you offer – consider adding a secondary site to ensure 100% service availability.
Then, perform the initial configuration of the system (typically there is an experienced PortaOne application engineer to assist you) to define your services, products and bundles. Each product is a combination of recurring fees, per-minute rate plan, bundled (free) minutes, limitations of concurrent calls (trunks) and per-second call attempts.
After that, provision customers in PortaSwitch by entering their authorization credentials (IP address or username / password), associated phone numbers and optionally applying anti-fraud measures. Add any value-added services such as call recording.
Finally, on your customer’s equipment, assign the high-availability PortaSwitch SBC IP address as the outgoing SIP proxy. Now any outgoing call from a customer’s office can be delivered to PortaSwitch. It is first authorized then routed to either another IP PBX (for intra-office calls) or out to a PSTN network. Calls made to DID numbers (either local or toll-free) arrive from carriers or DID exchanges and are then routed to the IP PBX which handles that number.
Now your customer has a clear overview of all the configuration settings, CDRs and invoices. At the end of each month all applicable fees are assessed, a PDF invoice is generated, emailed and the credit card that’s on file is automatically charged.
Additionally, you can increase the revenue by selling to more customers or by signing up the white-label providers (resellers and virtual operators) on your PortaSwitch platform.
The variety of IP telecom segments in which we operate demands a communications infrastructure that is not only reliable and scalable, but also able to respond to a constantly changing VoIP market landscape. With PortaBilling and PortaSIP, IXO and IziVox have both gained a reputation for innovative and proactive solutions. With these software platforms, PortaOne supports us with regular feature upgrades and exceptionally short time-to-market.”> Read more about IXO Systems success story
PortaSwitch as a SIP Trunking Solution – Benefits / Factsheet
Optimize daily operations
- Easy and quick service provisioning
- Anti-fraud and call attempts (CPS) control
- Automated billing and invoicing
- Detailed web-based logs for troubleshooting
Provide better experience for end-user
- Self-care to manage configuration
- Per-call selection of route quality
- Quick ordering of new DIDs (local, toll-free, international)
- Use desktop phone, mobile phone or smartphone app
Excel in service reliability
- Carrier-grade security with Oracle Enterprise Linux
- Full data redundancy
- Clustered components for high availability
- Ultimate availability (99.999%) with Oracle RAC
- Site redundancy and zero-downtime updates
- Real-time call authorization and overdraft protection
- Minute bundles with a fixed monthly fee
- Charging based on number of trunks and/or concurrent calls
- Least-cost routing across multiple outgoing carriers
Stay ahead of the competition
- Unlimited scalability with no per-user or per-feature costs
- Open platform that is quickly and easily expandable
- Fast delivery of new features with agile development
Rapidly increase your sales volume
- Online web signup
- Promotions and discounts
- Resellers (white-label operators)
- Usage/payment based commissions
- Least-cost-routing (LCR)
- Failover routing
- DoS attack and “rogue” call traffic prevention
- Multi-site office support (Voice VPN)
- Fixed-mobile convergence
- Static or dynamic IP address for branch IP PBX
- NAT traversal
- Local Number portability
- Fax: T.38 and fax-to-email
HD voice support
- On-demand DID provisioning
- Call limit control based on bandwidth
- SIP trunk control (total / outgoing / incoming)
- Call attempt rate (CPS) control
- Secure calling (SIP over TLS)
- Unified communication
- Customer self-care portal
- Automated invoicing
- Auto-charge of credit card / ACH
- VAS: Conferencing, CNAM, etc.
- API for creating advanced call control applications
- Statistics and reports
Toll-free calls (phone & Skype)
+1 866 747 8647
Calls & faxes from abroad:
+1 604 628 2508
Suite 408, 2963 Glen Drive
Coquitlam, BC, V3B 2P7