5 Key Points to Consider While Selecting an SIP Trunk Service Provider

Cloud telephony solutions are rapidly gaining popularity as businesses seek to upgrade existing infrastructures. Session Initiation Protocol (SIP) trunking is a process that allows businesses to replace physical phone lines by adopting virtual channels, which is a far more cost-effective and accessible replacement for traditional analog solutions. According to Globenewswire, the global market for SIP trunk services is expected to grow at a CAGR of 10.8% from 2020 to 2027.

SIP trunking has the potential to transform businesses by allowing them to update their PBX to an internet-based phone system. It is more trustworthy and helps in the optimization of business communications. SIP trunking is a cost-effective way to manage a phone system. It also makes it easier for businesses to add new phone lines or change their phone service without causing any delays or acquisition of additional hardware. Now, let’s understand SIP Trunking.

What is SIP Trunking?

SIP trunking is a web-based service that connects VoIP consumers with traditional phone networks. Rather than confining calls to voice alone, it enables Unified Communication, which includes video, web conferencing, and screen sharing, everything over a SIP-based Private Branch Exchange (IP-PBX).

A Session Border Controller (SBC) is commonly used by SIP trunking providers to manage communication across the SIP trunk and the company's phone system. An SBC enables secure and accurate data transfers across two networks while also improving traffic flow for better voice call quality.

Below is the list of top five features to look for before selecting a SIP Trunk Service Provider:

Top Five Must-Have Features for SIP Trunk Service Provider

Call Management

A SIP trunk consists of a variety of channels, each of which may manage one incoming and one outgoing call simultaneously. Therefore, businesses need to know how many simultaneous calls are manageable with the existing infrastructure to determine the appropriate number of channels.

Throughout the installation, most service providers can support evaluating call capacity requirements. SIP trunking is more flexible telephony solution than on-premises Primary Rate Interface (PRI), as it's much easier to grow the number of channels up with cloud-based SIP trunks.


Reliability is highly essential for voice communication and to get the standard Quality of Services (QoS). As voice signals may need a large amount of data, compression is used to minimize the bandwidth used for transmission yet preserving quality. For optimizing QoS and bandwidth, businesses prefer a provider that offers the effective codec such as G.711 or G.729 as per their business requirements.

It is also critical to find a service that connects the PBX to a dependable network using tier-1 carriers, to route calls to almost any network without incurring additional costs. The network must have multiple carriers for each location in which the business operates, to divert calls and avoid troublesome regions amid outages or other challenges. This enhances the uptime and availability of the telephony infrastructure.


As SIP trunks are used outside the company's network, it's important to ensure that the service provider complies with security and privacy industry standards. Besides analyzing the traffic and just accepting incoming traffic from the devices which are whitelisted within an Access Control List (ACL), an SBC might improve security. The SIP trunk can provide an added layer of security by just accepting the incoming traffic from managed provider’s SBC.


After ensuring that the PBX is SIP-compatible, it is useful to decide whether the PBX is interoperable with the SIP provider. Certain PBXs may not be compatible depending on the SIP platform offered by the preferred provider. Check the SIP interoperability list to ensure that the PBX is compatible. If a PBX is not labeled as compatible, organizations might have to engage some time in interoperability testing before installation.

It is probably the perfect chance to find out whether SIP can be used with non-PBX devices. Also, it might be possible that legacy devices like fax, alarms, and credit card machines were previously connected to the PBX through ISDN or PSTN connections. Make sure that the SIP provider can deliver these services over the new connection.

Direct Inward Dialing (DID)

SIP trunks are often assigned a variety of Direct Inward Dialing (DID) numbers that route incoming calls to specific SIP phones. Managed service providers might even support porting existing DID phone numbers to the new telephony infrastructure, ensuring that employees are not disturbed by the transfer.

Although existing numbers can be transferred in most cases, there are certain exceptions when the new DID numbers are purchased, such as when there is an absence of interconnection agreement between carriers or when the service has been terminated and the number has been issued to a new client. DID numbers are less expensive and more convenient to obtain, as SIP trunks don't need physical lines.

BridgeVoice is one of the leading players in providing inbound VoIP services. The platform allows termination of the call to a Direct Inward Dialing (DID) mapped to a specific number rather than navigating through a menu or an extension. The value-added service allows clients to easily develop, customize, launch, and manage comprehensive VoIP solutions. The solution also includes key features like voicemail, conference calling, automated greetings, and call logs to help our clients by cutting infrastructural costs.

Get The Best SIP Trunk Provider for Better Business Operations

Before SIP, workplace communications had separate voice and data communication frameworks which led to a communication and business continuity breakdown. Communications are now in sync with corporate activities thanks to total interoperability via VoIP and SIP trunking. Organizations now rapidly route and record calls, as well as manage emails making the communication more reliable and prompt with the use of video conferencing, instant messaging, and quick data transmission.

Naishil Jha

Naishil is a Content Writer at Panamax, Inc. with rich exposure in the field of Creative Content, Marketing Communications and Branding. With an academic background in Mass Communication and Journalism, he has made a career in content writing and has worked upon varied content pieces. In his leisure time he can be found reading about cricket, performing street photography and cooking some delicious food.