You have a lot of questions as customers, users, enthusiasts, and VoIP experts, and we’d love to answer them all. However, we are unable to respond to all of them. Not all at once, at any rate. So, to get started, the marketing department handed me a list of frequently searched VoIP queries that I can answer.
Is It Possible to Track VoIP Calls?
Yes. They are extremely traceable if the call must pass over the internet or the public switched telephone network at any point. You can’t do anything on the Internet without leaving some form of a digital trace.
Sure, you may surf using a “VPN,” which is simply a method of securely proxying your data to someone else’s server, but whoever runs that server has a record of all of that information.
And no one knows how secure their infrastructure is or with whom they share it. Carriers gather call data not only for the “greater good,” but also to improve service quality, monitor performance, and solve problems that emerge in the realm of hurling packets and multiplexed signals around the planet.
You can try to be untraceable by using a “security through obscurity” strategy, but that’s never what a professional does, and it’s actually simply a sign of illicit conduct.
To be honest, if you’re concerned about traceability, you’re missing the point. What you require is encryption. If you want to conduct a safety talk, do the following:
Do it in private, when no one else is present, or use SRTP and TLS with your own certificates. This should ideally be done using a VPN as well. OpenVPN, despite its name, maybe your best bet. It is still completely traceable in terms of source and destination, but your material (audio) encrypts, which is the most crucial aspect.
Where Do VoIP Packets Come From?
VoIP may employ a variety of protocols, but SIP is the current gold standard for setting up a conversation between two phones. In simple words, the caller sends an INVITE to the callee, who responds with a 200 OK. The caller then sends an ACK to complete the SIP three-way handshake.
Once the SIP handshake is complete, the media (RTP) between the caller and callee may begin.
When the call finishes, the person on the other end will send a BYE to notify the other party that the call is done. The call is subsequently “rip down” and the media stops pouring.
Why is VoIP superior to PSTN?
There are several reasons for this, but the reality is that whether routing a call to a mobile phone or traditional landline, the PSTN is virtually always involved.
When you call a mobile phone or traditional landline, the beauty of VoIP is most evident. When you make a call to another phone on the same network that may be in a different geographic location, the beauty of VoIP is most obvious.
For example, suppose you bring your VoIP phone with you to Singapore but need to call colleagues in New Orleans. You may link your phone (VPN capable phones) back to the office in New Orleans and dial an extension without incurring any international toll costs as long as you have access to a network in Singapore that supports a VPN connection outward.
It’s irrelevant that this can be done without a VPN, but I’d advise anyone utilizing a “foreign” network with any unencrypted communication to be cautious.
When will VoIP become the norm?
It has already happened. Did Verizon include a home phone with your FIOS service? That’s what VoIP is. Have you recently had a T1 installed at your workplace? That’s most likely VoIP.
Companies such as Verizon, Spectrum, and AT&T will deploy devices that convert SIP to a legacy interface such as an FXS or T1/PRI port. A SIP gateway is what it’s called. This enables you to utilize an analog phone or your PBX’s T1 port.
Carriers can route IP data much more cheaply using fiber optic networks rather than the old copper telephone infrastructure. It is still in use but is being phased out. VoIP isn’t only the future; it’s already here.
How does VoIP get routed?
A real VoIP call is routed with the rest of your Internet traffic after leaving your phone system. It’s no different from email or web packets.
What’s essential about VoIP in particular, and particularly the audio component of VoIP, is that it’s extremely time sensitive. You might not notice if a webpage takes an extra second or two to load, but that sort of latency in a discussion is quite obvious and can render a conversation incomprehensible.
Inside your PBX, it employs a dial scheme. A dial plan is a collection of conditional rules that directs incoming calls to an extension, ring group, call queue, or auto attendant. The dial plan routes calls depending on characteristics such as the number dialed, a specific caller ID, the time of day, or skills-based routing. Skills-based routing happens in a call queue, which provides queue agents a particular level of priority and may send more calls their way. The dial plan in most Asterisk-based phone systems is extremely changeable, making the system nearly indefinitely flexible.
