Build Better Global Connectivity through SIP Trunking


If you want to extend your connectivity beyond your country's borders, then you need to get behind global SIP. SIP trunking empowers you to grow your business across regions, so your outbound agent teams can connect with customers far and wide. 

It doesn’t matter where your offices are physically located, or where you’re calling to– SIP enables end-to-end connectivity. It’s a technical protocol that lifts traditional telephony onto the internet and delivers scalable, high-quality voice services. 

Global SIP benefits make trunking an excellent solution to increase connectivity. Below are its components, which you can easily use to support and expand your contact center’s voice network.

SIP Calling

Also known as SIP voice, this is the process of transmitting phone calls via a SIP trunk or channel using VoIP. Its capabilities move your contact center’s voice system to the cloud, so you can do more with your legacy infrastructure. 

SIP Trunks vs Primary Rate Interface (PRI) Lines

Although SIP trunks and PRI lines serve a similar communication purpose, their methodologies are quite different. 

Phone calls placed using SIP trunks are completed over an internet connection, whereas calls made via a PRI are completed over traditional copper lines using the Public Switched Telephone Network (PSTN). 

There are quite a few obstacles to calls made using PRI and PSTN, including poor call quality and reliability and voice coverage limitations. With the cloud’s direct ties to the internet, its capabilities create seamless connections worldwide, providing you the experience of high-quality and reliable calls wherever your customers are without the need for costly on-prem infrastructure.

SIP Protocols

Using SIP in your voice network means you can be confident in your global connectivity and call quality. Global SIP trunking uses UDP, TCP and TLS layered protocols to ensure call security and encryption during the transport phase. 

Protocol Benefits
User Datagram Protocol (UDP) Rapidly moves condensed voice and media data packets between hosts, only providing the essentials required.
  • Speed
  • Efficiency
Transport Control Protocol (TCP) This protocol type delivers data packets in a particular order to provide users with reliable messaging and voice delivery via the internet.
  • Reliability
Transport Layer Security (TLS) Uses encryption to ensure end-to-end security.
  • High-security requirements
  • Authentication
  • Data integrity

Consider your security and delivery requirements when deciding on the best type of SIP protocol for your business. UDP is the industry standard, but many global SIP providers offer TCP and TLS to accommodate personal preferences, local restrictions and business needs. 

VoIP Codecs

A VoIP codec is an algorithm that compresses audio data transmitted during VoIP calls to improve their quality and lower bandwidth requirements. 

In simple terms, VoIP codecs work to provide faster, higher-quality global connectivity for businesses. 

There are several types of codecs offering different rates of compression, quality, and bandwidth. Three common types of VoIP codecs include G.711, G.722 HD, and G.729.

Codec Name Disadvantages
  • Can use for most VoIP applications
  • No licensing fees
  • No digital decompression
  • Simple to employ
  • High bandwidth requirements
  • Lack of support for multiple phone calls at once
  • Toll-quality audio
G.722 HD
  • High definition, wideband
  • Improves speech quality
  • Decreases latency
  • High compression rates
  • Large audio frequency range
  • Not as extensively supported as other G series codecs
  • Unsuitable for WANs
  • Low bandwidth requirements
  • High compression rates
  • Support multiple calls at once
  • Not all VoIP providers support it
  • Non-verbal audio and media may experience poor quality
  • Licensing fee requirement

Though we mentioned some of the more popular types above, there are many other VoIP options you can explore for your business needs. 


A User Agent (UA) is a VoIP-based phone acting on behalf of users as a network endpoint. These can be…

  1. IP-enabled desk phones
  2. Softphones

Two potential UA communication processes are User Agent Client (UAC) and User Agent Server (UAS). 

UAC serves as the starting point, sending SIP requests out. An example of this is sending a request to accept a phone call. 

In turn, UAS serves as the ending point, receiving the SIP requests from the UAC. It returns a reply in the form of accepting, rejecting or setting parameters for that call. 

SIP Gateway

Plugging a SIP gateway into your existing infrastructure improves global connectivity by converting traditional phone signals to IP-based phone signals. These gateways can be either hardware or software, creating a connection between networks so data can flow from one to another securely and reliably– important factors of your business’s quality of service (QoS). 

Quality of Service (QoS) Router Setting

This is a router setting within your SIP gateway that communicates with the network to prioritize voice calls over other traffic. Why does this matter? QoS router setting helps reduce the lag between calls, allowing you to sustain high audio quality while simultaneously working through other data-intensive activities. 

Tier Carriers

Your customers expect excellent audio quality and phone calls that don’t drop. Therefore, it’s essential to select a global SIP provider that offers the best-grade carrier service. 

Those carriers are Tier-1 carriers, which offer the following benefits: 

  1. Superior infrastructure. Tier-1 carriers often have large financial resources and tons of infrastructure directly connected to the backbone of the internet for high-quality calls. 
  2. Reliability. They offer more points of presence (PoPs) and carry traffic further using cold potato routing. 
  3. Greatest global reach. With a large footprint and coverage areas, reach as many customers as possible with fewer obstacles and resistance.  

Tier-1 carriers will ultimately help you provide your voice network and contact center teams the connectivity needed to keep pace with your business’s outbound calling objectives. 

But what about Tier-2 and Tier-3 carriers? These carriers must rely on IP transit purchasing permission from Tier-1s to obtain access to the internet. Because of this, businesses using Tier 2s and below are more likely to experience lower reliability and are subject to service changes. 

Quality of Service

Global SIP providers should offer several ways to measure and monitor your network call traffic, application performance and account security. The quality of service you receive goes hand-in-hand with call quality and connectivity. 

You need someone in your corner, a partner with robust SIP expertise– so you can deliver seamless communications to your contact center’s voice network. Consider a provider with built-in support features, such as:

  1. Troubleshoot jitter, packet loss and poor MOS/PESQ
  2. Account activity monitoring
  3. 24/7 call traffic monitoring
  4. Fraud prevention measures
  5. Premium-level support 

Increase Outbound Call Connectivity with Global SIP Trunks

Now that you’ve reviewed all of the components involved in SIP trunking, are you ready to increase your global connectivity? Don’t leave your SIP calling to chance with just any provider— while you have tons of options, consider a global SIP provider who will be your partner in helping you stay connected across the world. 

The AVOXI Platform offers several SIP integrations you can use to improve your global connectivity. Use our voice services to simplify and manage your SIP trunks, global phone numbers, documentation and more in one place. See how in our on-demand voice demo. 

Sneak Peek Demo

Enterprise-Ready Global Communications Platform

Watch our 9-minute video for a brief look at how you can configure, customize and gain more reach using global SIP trunking on the AVOXI Platform.