Taking a look at the basics of the SS7 Protocol Stack.
Some tricks to handle if you’ve got multiple headers all with the same name in Kamailio
Through fs_cli you can orignate calls from FreeSWITCH. At the CLI you can use the originate command to start a call, this can be used for everything from scheduled wake up calls, outbound call centers, to war dialing. For example, what I’m using: originate sofia/external/[email protected]:5061 61399999995 XML default originate is the command on the FS_CLI […]
Using the Rtimer module and UAC to be annoying, or very useful.
A glimpse into the complexities of prepaid billing (Online Charging) Diameter based networks.
Using mod_httpapi to HTTP POST call recordings to a remote server
Using ENUM to resolve E.164 numbers into SIP URIs using Kamailio.
A Bind environment in a Docker container for experimenting and learning ENUM for DNS based Call Routing.
Using DNS to resolve E.164 phone numbers to routable SIP URIs
Peer behind the magic curtain at how IMS networks route your VoLTE and VoNR calls, and how iFC (Initial Filter Criteria) achieve this.
Adding support for AMR Codec in FreeSWITCH
I’d been trying for some time to get Kamailio acting as a Diameter Routing Agent with mixed success, and eventually got it working, after a few changes to the codebase of the ims_diameter_server module. It is rather unstable, in that if it fails to dispatch to a Diameter peer, the whole thing comes crumbling down, […]
It’s 2021, and everyone loves Containers; Docker & Kubernetes are changing how software is developed, deployed and scaled. And yet so much of the Telco world still uses bare metal servers and dedicated hardware for processing. So why not use Containers or VMs more for VoIP applications? Disclaimer – When I’m talking VoIP about VoIP […]
Using Docker to spin up environments to test Kamailio in
So far with most of our discussions about Kamailio we’ve talked about routing the initial SIP request (INVITE, REGISTER, SUBSCRIBE, etc), but SIP is not a one-message protocol, there’s a whole series of SIP messages that go into a SIP Dialog. Sure the call may start with an INVITE, but there’s the 180 RINGING, the […]
The buttinski test phones I know and love.
How SIP hold using RFC6336 is implemented and how it looks in production.
Using FreeSWITCH to serve WebSocket / WebRTC connections
Things to keep in mind before using Kamailio as a Load Balancer for Asterisk
Kamailio world was an online event this year, but you can find all the videos here now they’ve all been posted.