If you take one thing away from this thread, it should be learning how to take and read packet captures. Most problems in VoIP are plainly obvious given one or two packet captures, especially if the problem is network related. Know how to capture packets at every hop you have control over. Sometimes the problem is in between two elements under your control and packet captures from both sides can tell you what’s going wrong.
Also being able to read packet captures makes understanding all the other parts a lot more straightforward.
SIP is obviously the primary protocol you’re going to want to get familiar with. Understand how registration works, how a call is set up, and how it interacts with RTP. Most of SIP is human readable plain text very similar to and often based on HTTP, so a basic understanding of HTTP is also useful to have.
There’s not a lot to understand about RTP itself, but understanding how RTP flows are set up by SIP messages is key. Beyond that, just understand how to use your tools to decode captured RTP in case you need to diagnose call quality issues.
If any of your phones go over the IPv4 internet you’ll also want to have a good idea how NAT works, particularly source port randomization as this can cause many headaches if one end of the session doesn’t understand there’s a NAT layer involved.
Beyond those three things, in no particular order:
DHCP - Almost every phone will depend on DHCP for its addressing, and in a business environment DHCP will likely also be used to point the phone at the config server.
VLANs - Most larger deployments will use VLANs, have a basic idea how they work.
LLDP - If using VLANs, this is the standard way to inform the phone what VLAN it should be using.
TFTP - Most modern devices support HTTP configuration, but TFTP is still around and you’ll probably encounter it from time to time.
DNS - Insert DNS haiku here.
NTP - It’s shocking how often I have phones that can’t set their clock.
You shouldn’t need a ton of depth on any of those topics, but you should understand what the basic flow looks like so you can tell when it has gone wrong.
If you take one thing away from this thread, it should be learning how to take and read packet captures. Most problems in VoIP are plainly obvious given one or two packet captures, especially if the problem is network related. Know how to capture packets at every hop you have control over. Sometimes the problem is in between two elements under your control and packet captures from both sides can tell you what’s going wrong.
Also being able to read packet captures makes understanding all the other parts a lot more straightforward.
SIP is obviously the primary protocol you’re going to want to get familiar with. Understand how registration works, how a call is set up, and how it interacts with RTP. Most of SIP is human readable plain text very similar to and often based on HTTP, so a basic understanding of HTTP is also useful to have.
There’s not a lot to understand about RTP itself, but understanding how RTP flows are set up by SIP messages is key. Beyond that, just understand how to use your tools to decode captured RTP in case you need to diagnose call quality issues.
If any of your phones go over the IPv4 internet you’ll also want to have a good idea how NAT works, particularly source port randomization as this can cause many headaches if one end of the session doesn’t understand there’s a NAT layer involved.
Beyond those three things, in no particular order:
You shouldn’t need a ton of depth on any of those topics, but you should understand what the basic flow looks like so you can tell when it has gone wrong.