Have you ever wondered how the internet differentiates between different types of data; digital voice signals, websites, emails and so on? The answer can be as simple as how the post office works and deliver different types of parcels to people. It’s been done using ports’ numbers.
If you want to learn more, please visit our website 4 Ports Series VoIP Gateway.
Every single data from the outside world has a destination port attached to it, exactly like how parcels are categorised to boxes, mail, fragile items, etc. This makes it easy for the server to sort its traffic and deliver data to the correct destination (the service). There are 65535 possible ports, so a server can have many different services listening on different ports. For example, web servers traditionally listen on port 80 for http and port 443 for https.
VoIP traffic is no different in that; a VoIP server must listen in on a port for its incoming traffic from VoIP phones and the upstream telco.
The industry standard port for VoIP traffic is 5060. Usually when connecting your desk phone to a server, the port is already set to 5060 by default so you don’t need to enter it.
What port a server listens to is not set in stone though. Many people choose to change from port 5060 to a random different port. This has the advantage of avoiding traffic from malicious bots attempting to attack VoIP servers. These bots are nearly always designed to probe for servers on port 5060, so a server listening on a different port will not see this traffic.
The other advantage to using a non-standard port is that you can have many VoIP servers on one piece of hardware, all listening on different ports for different customers. This reduces costs and makes more efficient use of the server resources.
A reseller for VIP VoIP needs to know the port the server is listening on so that they can configure the VoIP phones for new customers.
VoIP traffic comes in two types:
1- Control traffic; also known as Session Initiation Protocol (SIP). This traffic is used for registering extensions to a server, making and receiving phone calls, subscribing to voice mail updates etc. Control traffic does not contain any audio.
2- Audio traffic. This traffic is purely for audio. Once setting up a call is done via SIP, then both sides begin sending audio traffic on separate ports to the SIP traffic. A single VoIP call will need four ports at the server. Two for receiving and two for sending to each side of the conversation.
SIP traffic generally tends to use the User Datagram network protocol (UDP) to traverse the internet. UDP has the advantage of being light on resources as it is connection-less. With this protocol, there is little overhead in error checking and servers and clients do not have to negotiate connections with each other in order to send traffic.
UDP has some distinct disadvantages though. For one there is very little error checking performed, it is possible for traffic to be corrupted in transit, or simply not arrive at all. All VoIP UDP traffic is sent in plain text as well. Anyone eaves dropping on the traffic can recover passwords easily.
We recommend using Transmission Control Protocol (TCP) with TLS to connect phones to a VoIP server. TCP has the advantage of being connection based, with good error checking and retransmission. This makes control traffic (such as notifying a phone of an incoming call) between server and client much more reliable. The addition of TLS encrypts the traffic making eaves dropping incredibly difficult.
If you’d like to find out more about VIP VoIP, how it works and whether it’s the right system for your business, then get in touch with us today and we’ll do our best to give you an honest advice.
You can call us on 03300881182 or send us an email at [email protected]
If you are looking for more details, kindly visit 16 Ports SMS Gateway Modem.
Hey there,
first of all make sure that in your “modem” all options for VOIP are “unblocked”. I have a FritzBox and once I accidentally blocked VOIP in the FritzBox, then no matter what I tried on the IPFire, calls would never go through.
Anyway, concerning IPFire:
Clarify if you need SIP (Port 5060) or SIPoverTLS (Port 5061), it depends on wether your phone system uses encrypted connection or not.
Simplified: The SIP-ports are for building up and managing the call itself, think of it as the “permission to ring”.
10000-20000 look like RTP-ports. These ports are needed for the audio transmission, as soon as the call is answered. In IPFire you need to use a colon if you want to define a port range. “10000:20000” would mean every port from 10000 to 20000.
443 obviously is https, I guess this is for your voicevox to connect to a server? I don’t know how these voicevox works, so I can only guess BUT because you wrote that you need to open the ports to only two external IP adresses, I would assume that the voicevox is something like a client, that connects to a server on the web.
In this case, you only need to allow OUTGOING connection, so from GREEN (or rather the voicevox) to the external IP-adresses.
Depending on your setup (what is between your IPFire and the Internet?) and the configuration of your IPFire in general (NAT activated?), you do not need to activate NAT specifically in this one firewall-rule.
Wrapping up (the comfortable way):
Create a host: 216.198.100.250
Create another host: 216.198.105.90
Create a host-group with both hosts in it.
Create a service for SIP (5060) or SIPoverTLS (5061).
Create another service for RTP (10000:20000).
Create another service for HTTPS (443).
Create a service-group with all the services in it.
Allow traffic from your voicevox to the newly created host-group and as protocols choose the newly created service-group.
Personal opinion: You could separate the traffic to the two different external IP addresses for security reasons but this seems a bit overkill in this case.
Please note: If my assumption is wrong and the voicevox is something like a phone server itself the RTP and SIP ports have be opened to the whole RED network and not just one IP because these ports must be opened to every IP address you want to call (at least for “direct connections”). On the other hand, there would be no need for the open port 443 to an external IP address if the voicevox is a server itself. Also if the voicevox is a server you need to allow INGOING traffic as well because it must be reachable for people who want to call you.
A bit much text but I hope this explains some things. =)
Greetings
Alex
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