Cisco SIP SRST Configuration is the process of providing backup to external SIP Call control (IP-PBX) through the provision of the basic registrar and re-direct server or back-to-back user agent (B2BUA) services.
Configuring options Ping on CUBE, and learning how to busy out a dial-peer when a SIP trunk goes down, are essential skills that UC engineers are expected to be good at. In this blog post, we will explore the different SIP options available.
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 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!
Connecting the Cisco IOS Voice Gateway to CUCM via SIP has been the preferred way to do it in the past couple of years.
The slowly dying H323 protocol (ISDN based) is not being developed anymore while SIP (HTTP based) became the industry standard for VoIP. So whenever PSTN connection is implemented via an IOS Voice Gateway, the choice should be really between SIP & MGCP.
There 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.
Why? Easy. 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.