Smart solutions to VoIP questions
Introduction This is the final article in my series about fixing SIP header addresses. It broadly overlaps with the theme of the earlier articles. As a VoIP solutions designer, you may want your proxy server to deliberately transcode between different transport protocols. For example, WebRTC to TCP or TLS to UDP. This is possible with … Read more
Part 3 of this series of articles focusses on the Contact header. In particular, I examine the use-cases where it is necessary to “fix” (or alter) a received Contact header. Contact headers work in close combination with Record-Route and Route headers in a mechanism known as loose routing. To get the most from this article … Read more
In part 2, Via headers are put under the microscope. I examine how the address in the Via header is set by each node in the path; how and why it may differ from the source address. I will look at the functions available in OpenSIPS to detect and handle situations where the address in … Read more
The main theme I explore in these articles is when and how a SIP Proxy should alter (or “fix”) embedded sender address information – IP and port – in a SIP request that it has received. The headers that are most relevant here are Via, Contact and Record-Route
Diagnosing some problems in the world of VoIP requires close inspection of the SIP messages being exchanged, but there are many occasions where a good understanding of loose routing will be invaluable. The headers that underpin loose routing are Contact, Record-Route and Route. In this post, I explain how they work and provide some insight … Read more
VoIP based phone systems bring many benefits, but they also bring some problems. Not least is the annoying tendency for some calls to drop mid-way through your conversation for no obvious reason. In this article I will identify the most common reasons why a VoIP call might suddenly drop mid-way through an established call and explain … Read more
In this article we will briefly look at what RTP is and how it is used to stream VoIP audio. The article then considers how certain network transmission characteristics may introduce jitter or packet loss and the measures that are used in VoIP equipment to mitigate the effects. Other phenomenon which have a bearing on … Read more
Manufacturers different naming conventions While SIP is an industry standard protocol, the names assigned to the configuration fields on an IP phone are far from standardised. The tables below show the text labels that you can expect to see on the web interface configuration forms next to the boxes where you must enter your settings. … Read more
SIP Devices behind NAT: What solutions are available? When an IP phone is installed behind NAT, problems can be created by the NAT device itself, by the phone’s inability to correctly understand its own networking environment or from a combination of the two. Because it is such a common problem, most IP Phones have built-in … Read more
Why is it a problem using SIP Clients behind NAT? What is NAT? To understand why SIP Clients behind NAT are a problem, you need to first have some understanding of what NAT is and what it does. NAT stands for Network Address Translation. Unless you are using One-to-one NAT, then a NAT device may … Read more