The Three Paths to VoIP: SIP Trunk, SIP Registrar, and IAX
· Callr
Whether you run a contact center or want to build telephony into your own product, business VoIP can feel like a maze. The setup that works for you depends on several factors at once: the quality of your internet connection, your available bandwidth, how your network assigns IP addresses, how many simultaneous calls you need, and which codecs you use. There are three ways to connect: SIP Trunk, SIP Registrar, and IAX. Each one carries its own trade-offs.
If you have never done this before, the choice can look intimidating. It does not have to be. This guide walks you through the variables in order, from your internet connection up to your telephony needs, so you can land on a setup that genuinely fits. We will start with the fundamentals, then build toward a clear recommendation.
What is VoIP?
Telecom, and VoIP in particular, is full of acronyms that can put newcomers off. We will keep things plain even if you are not an expert.
VoIP is a group of technologies that deliver voice and multimedia communication over the internet. Instead of relying on the Public Switched Telephone Network (PSTN) that carries traditional landline and mobile calls, VoIP uses IP networks, which brings several advantages. It can run over the public internet or inside a private network, such as a company LAN.
The voice signal is digitized and split into packets, which makes transmission far more efficient. Because the audio is digital, several calls — what we call “channels” — can share the same line at once. In the legacy phone system, one line essentially meant one call. VoIP needs either dedicated hardware (a VoIP phone) or software, and it is the foundation under many everyday apps for voice and video calling.
Since VoIP rides on the internet, the quality and characteristics of your connection are the single biggest factor in call quality. That is where we go next.
Capacity planning and sizing
Bandwidth — the maximum transfer rate of your network — ultimately sets how many channels (simultaneous calls) your setup can handle. But bandwidth is not the whole story. Depending on your configuration, you can compress the signal to reduce the bandwidth each channel needs, which lets you fit more channels into the same connection.
To size your installation correctly, you need two numbers:
- The maximum bandwidth and ping of your internet connection. This part is straightforward.
- The codec you will use, since it determines how much bandwidth each call consumes. This takes a decision on your part.
Together, those two figures tell you how many channels your setup can support.
Measuring your bandwidth and ping
Free online tools can measure the maximum upload and download speed of your connection along with its response time (ping). Any reputable speed-test service will do.
Your upload speed is usually the limiting factor, because most consumer connections are weighted toward download. Ping is the time a packet needs for a round trip. On its own, ping is not the main driver of call quality — what matters more is its stability. Ping spikes are what you hear during a conversation. As a rule of thumb, aim for a ping below 40 ms. Above 120 ms you will start to notice the delay, and above 160 ms the line becomes choppy.
One caveat: internet service providers often optimize for these tests, so results reflect a best-case maximum rather than the speed you can count on day to day.
Codecs: what they are and what they do
Sending an audio stream — your voice, for instance — over a digital medium means converting an analog signal (sound in the air) into a digital one (your voice encoded as ones and zeros).
What codecs do
A codec handles both the encoding and the decoding of the voice signal. If the caller uses a given codec, the recipient needs the same codec to decode it. Beyond transmission, a codec can also compress the data to shrink its size and lower the bandwidth each call needs — but compression costs quality.
There are two broad families. “Lossy” codecs compress aggressively; “lossless” codecs prioritize preserving the original quality. Each codec is optimized for a specific use. Voice calls demand very low latency between encoding at the source and playback at the other end, so telephone codecs are typically lossy: they cut bandwidth and keep latency minimal at the expense of audio fidelity. That is the origin of the familiar “telephone voice.”
The codecs Callr supports
Callr relies mainly on two codecs. Both are widely used across the telecom industry, which keeps interconnection smooth.
- G.711 / PCM (lossless): PCM delivers very high quality, but it is uncompressed and needs a lot of bandwidth. It comes in two slightly different variants — μ-law (PCMU), used in North America, and A-law (PCMA), used elsewhere.
- G.729 (lossy): G.729 is built to reduce the bandwidth an audio signal needs. It compresses the stream heavily while keeping quality good enough for phone calls.
Your bandwidth requirement follows directly from your codec choice. G.711, being lossless, is demanding: a standard ADSL line might support only 10 to 12 channels. Because G.729 compresses the signal, the same line could carry up to roughly 100 channels.
| Codec | Type | Quality | Bandwidth per channel | Channels on a standard ADSL line |
|---|---|---|---|---|
| G.711 / PCM | Lossless (uncompressed) | Very high | High | ~10–12 |
| G.729 | Lossy (compressed) | Good (phone quality) | Low | ~100 |
Authentication: dynamic vs. static IP addresses
The way your network assigns IP addresses is another deciding factor between SIP Trunk, SIP Registrar, and IAX. It has no bearing on the number of channels you can run, but it is central to authentication. Before we get into the options for each network type, a quick refresher.
What IP addresses are and why they matter for VoIP
Every device on the internet is assigned an IP address, which may or may not be unique to that device. There are two assignment modes:
- Static IP address: a fixed address tied to a given device. It stays the same even after the router or device reboots. Because it is constant and unique, it can be used to authenticate its owner.
- Dynamic IP address: the network assigns an address each time the device connects. Because it changes over time, it cannot be relied on for authentication.
Other mechanisms can also change the address your traffic appears to come from — Network Address Translation (NAT) being the most common — which affects authentication too. We cover NAT below, but first, why IP matters for SIP Trunk authentication.
Authentication by IP address
Callr SIP Trunk authenticates by IP, so it will not work with a dynamic IP — you need a static address. IAX and SIP Registrar, by contrast, use the Password Authentication Protocol, so they work fine on a dynamic IP.
Where NAT and VoIP collide
NAT hides private IP addresses behind a single public one, letting many devices on a LAN share that one public address. All outgoing traffic passes through a NAT router, which swaps the source IP for its own public address.
SIP Trunk uses two separate data streams — one for signaling and one for the audio payload — and the port carrying the audio is chosen at random. So even when the NAT router handles the signaling, it has no way of knowing which audio stream that signaling is paired with, or where to send it. From the user’s point of view, the result is one-way audio or no audio at all, despite signaling that looks healthy: the phone rings, the caller hears the ringback, the called party sees the caller ID, the ringing stops on pickup — but there is no voice.
Because IAX carries both signaling and audio in a single stream, it sidesteps the NAT problem entirely.
Reaching Callr’s infrastructure
You can reach Callr through two entry points. It is good practice to use both in round-robin, splitting your traffic across them to reduce the risk of an outage taking your installation down.
Firewalls
A firewall controls traffic against a set of security rules, usually a whitelist of approved IPs so that authorized applications keep their access. If you restrict inbound or outbound traffic, you will need to allow the right addresses and ports for Callr’s SIP service to work. The specific IP ranges and ports to whitelist — including the RTP port range and the SIP signaling port — are documented in the Callr Console and our developer docs; configure your firewall to match.
The three paths to VoIP
With the underlying technology and its challenges in hand, we can compare the three implementations on equal footing.
The SIP protocol
SIP is the industry-standard protocol. A wide range of manufacturers support it, and its request-response structure is easy to work with: a client sends a request, the server replies with a response that always begins with a status code. That model will feel familiar to most developers, because SIP reuses much of the HTTP request/response model, including its status codes. As a result, SIP installations are relatively easy to debug.
As noted earlier, voice in VoIP is “packetized” — bundled into packets. A SIP call exchanges two kinds:
- Signaling packets, which set up the call (for example, negotiating a shared codec) and keep it alive.
- Media packets, which carry the conversation itself.
SIP Trunk gives you full control over your phone system. SIP Registrar is a more accessible on-ramp to SIP: it works with softphones, which makes the initial setup considerably simpler.
SIP Trunk
SIP Trunk is the industry standard and generally delivers the best call quality, with broad gateway compatibility. It requires a static IP, since the IP is used for authentication. That reliance on IP means SIP Trunk can run into trouble behind NAT.
SIP Registrar
SIP Registrar is aimed at the end user and comes with many built-in features — some gateways even require it. It is more constrained than SIP Trunk, though: a few features, such as custom caller line identification (CLI), are not available on Callr SIP Registrar. Like SIP Trunk, it can have issues with NAT.
IAX
IAX (Inter-Asterisk eXchange) is a different approach altogether, with clear advantages on low-bandwidth connections. It uses a single stream for both signaling and media, which makes it far less resource-hungry. It authenticates with the Password Authentication Protocol rather than by IP, so it works on a dynamic IP and handles NAT and firewalls cleanly. Callr’s IAX setup also ships with a full set of codecs, which makes international calling easier and saves you the effort of configuring them yourself.
The main constraint is that IAX is built on Asterisk — an open-source telephony platform — so you must use an Asterisk gateway to run it. Overall, IAX is a strong alternative to SIP Trunk for installations with complex network setups.
Finding your VoIP fit
Here is the short version to guide your decision:
- Static IP, want the best call quality and full control? Choose SIP Trunk.
- Want a quick, softphone-based setup, or your gateway requires it? Choose SIP Registrar — just note its feature limits, such as no custom CLI.
- Dynamic IP, behind NAT, limited bandwidth, or running Asterisk? Choose IAX.
We hope this helps you settle on a VoIP implementation that fits your needs. If any questions remain, our team is happy to help — get in touch and we will walk through it with you.