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For miniSIPServer V60, the lowest requirement of Debian is changed to the “oldoldstable” version which is Debian V10 at this time. That means Debian V8 and V9 will not be supported now.
Of course, the lowest requirement of Ubuntu is also changed to Ubuntu V18.04.
Please refer to the document for more details about Linux systems.
miniSIPServer is upgraded to V60 which is the latest stable version for business development. The first big thing is “conference room” feature which provides conference calls for local users. At most 5 clients can join the same conference call. Please refer to the service document for more details. Cloud miniSIPServer is also upgraded to support this feature.
In another way, as we have posted in previous blog, several services are finally removed from local miniSIPServer, such as calling-card and call-shop. These features were important for some of our customers, but it is time to say good-bye now.
As we know, miniSIPServer was developed about 20 years ago. Lots of services and features are added into miniSIPServer to support more and more customers.
Recently we have reviewed all these services. Some services have so long history that we have to think whether they are suitable for current environments, for example call-shop, calling card, and so on.
Next version will focus on refining or clearing some services. miniSIPServer will step into next stage and be more faster, more stabler.
Ubuntu 24.04 is the latest LTS (long-term support) version, so it will be deployed widely in business environment. We install miniSIPServer on this important version and make some tests. The result is perfect! Please refer to the figure below.
If you want to deploy a new VoIP network on Linux system, Ubuntu 24.04 could be a good choice.
Please refer to online document for more details about how to run miniSIPServer on Linux system.
miniSIPServer recently is upgraded to support TLSv1.3. This modification doesn’t affect configuration, so you need to do nothing if you upgrade your miniSIPServer to the latest versions.
Two modules could use TLS transport: (1) SIP server, and (2) Embeded HTTP server. If your SIP phones can support TLSv1.3, it is better to use TLSv1.3 to protect communication. Please refer to “SIP over TLS” document for more details. Both local miniSIPServer and cloud miniSIPServer can support SIP over TLSv1.3 now.
By default, miniSIPServer starts an embeded HTTP server for web management. If you want to manage it through the pubilc network, you have to enable TLS transport to protect HTTP information. In another way, most navigators, such as Chrome, Edge, Firefox and so on, can support TLSv1.3 now. Please refer to “web management” document to enable encrypted HTTP.
As we know, miniSIPServer has some versions for Raspberry Pi and they are all for armhf architecture. Recently, more and more customers ask us for miniSIPServer versions for ARM systems. Most are arm64 architecture, and the customers want to run miniSIPServer on ARM servers or cards.
So we change the specific miniSIPServer version for Pi to the common miniSIPServer version for ARM64. Of course, raspberry pi can support arm64 architecture too, so this modification can cover most ARM scenarios and devices, including Pi.
In another way, most customers want to run miniSIPServer command line version on their ARM servers or systems. That means it is unnecessary for them to have a GUI interface, and they only need ‘minisipserver-cli’. By default, miniSIPServer requires ‘qtbase5-dev’ package to provide GUI. In this scenario, the ‘qtbase5-dev’ package will not be necessary, so we move this package from ‘Depends’ section to ‘Suggests’ section of miniSIPServer’s deb-control.
If you want to run miniSIPServer with GUI, you can still install the libraries with the following command:
sudo apt install gcc g++ qtbase5-dev
If you only need a command line version, you can install the libraries without qtbase5-dev, like following:
sudo apt install gcc g++
“Call forwarding” is a very traditional service in VoIP or communication fields. By default, SIP clients can send 3xx messages to miniSIPServer to invoke a forwarding. In another way, miniSIPServer can also directly invoke forwarding by itself.
But when the callee side is being forwarding, the caller side knows nothing about it. In most scenarios, the caller parties don’t care the forwarding. but some customers sometimes need to know what happens when the call is being forwarded.
miniSIPServer can send 181 “Call Is Being Forwarded” messages back to the caller side to update it that callee side is being forwarding. In the 181 messages, miniSIPServer will add a Call-Info header to indicate the forwarding information. Please refer to the figure below.
In this figure, there are two forwardings, (1) user B is being forwarded to user C; and (2) user C is being forwarded to user D.
The Call-Info header of the 181 message will indicate (1) the call is being forwarded, (2) who is being forwarded, and (3) who is being forwarded to. Please refer to the Call-Info header of the first 181 message which indicates user B is being forwarded to user C.
Call-Info: purpose=forwarding, username="userb", contact="userc"When miniSIPServer sends out SIP messages, such as REGISTER or INVITE messages, to VoIP providers, it will add a parameter ‘user=phone’ after Request-URI. It is required by China Mobile network. In most scenarios, it is no problem since URI parameters are defined in RFC3261.
But unfortunately, some customers recently reported us that their miniSIPServers failed to connect to their VoIP providers because these providers’ servers cannot recognize parameters of Request-URI. Of course, the easy way is that the VoIP providers upgrade their servers to fit RFC3261, then everybody will be comfortable.
Some of them insist on their status and refuse to make any change. Then we have to make a change in external lines’ configuration. Please refer to the figure below.
We add a new item “additional parameter of Request URI” in external lines’ outgoing call configuration. Then customers can control the parameter according to their real network environments.
In another way, if the GUI is in Chinese which means the customer might configure miniSIPServer for China networks, the default value of such item will be “user=phone”. Otherwise, its default value is blank. We think it will flexibly fit the network requirements around the world.
Debian 12 (bookworm) was released. It is the latest stable version and will be widely deployed in business environment absolutely. So we run and test the latest miniSIPServer on this system as usual. Of course, the result is perfect.
Please refer to the figure below.
If you want to build a VoIP system on Linux system, Debian 12 is a good choice.
Please refer to our online document for more details about how to install and run miniSIPServer on Debian systems. And I’m sure you’ll like the combination of Debian and miniSIPServer.
We upgrade web UI for miniSIPServer, including cloud miniSIPServer and local miniSIPServer. The new web UI is quite like GUI of local miniSIPServer. Please refer to the figure below.
We hope users who are familiar with local miniSIPServers can enjoy it and experience the cloud miniSIPServer system quickly.