Using Wireshark SIP Analysis for VoIP scenarios

Troubleshooting VoIP issues can be troublesome. When something goes wrong, the lack of direct visibility as to what is occurring on the network with SIP and RTP packets can initially be intimidating to network and voice engineers. However, Wireshark SIP analysis turns ordinary engineers into superheroes, allowing them to see deep into the network and determine exactly what is happening.

In a series of previous articles, we took a look at several methods that can be used to capture voice packets on a network. We also introduced the Wireshark packet sniffing software, providing you with a solid foundation for understanding how captured packets are obtained and stored.

In this article, we’ll get our hands dirty by examining a real voice packet capture from a production network. We’ll go through the whole scenario, delving deeply into the details of the voice packets being exchanged. Let’s get started!

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CUCM SIP Call Flow Troubleshooting

SIP Call FlowThere is no way around it,
finding the SIP call flow is the first thing you have to do when you are facing a SIP call failure.

Because first of all, you have to understand whether this is a call routing problem or signaling/media compatibility issue.
SIP call flow helps you understand just that, and in a lot of cases, you can pinpoint the problem just from looking at the SIP call flow.

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