We updated miniSIPServer to fit 4K screen recently, but we find that we have to upgrade our tool chains at the same time if we want to get a perfect result.
Unfortunately the new tool chains cannot support Windows XP/2003 systems, so it is time to say good bye and move on now.
The latest version (V40) is released yesterday and it requires Windows 7 or abover version if you want to deploy miniSIPServer on Windows system. V40 is also rebuilt for Debian / Ubuntu systems to fit higher DPI screens.
Please enjoy the new versions. And please update us if you have any questions or opinions. Thanks.
As we know, miniSIPServer can run on Linux systems. Sometimes customers ask us about SIP clients which can run on Linux systems at the same time. In fact, there are lots of choices, such as linphone, jami, and so on.
Recently there is a new SIP client released. Most important, it is a core application in Gnome project. It is “Calls”. In Debian library, its definition is “Make and receive PSTN phone calls”. In fact, the latest version can support SIP protocol. In the Gnome project, we can see the description is changed to “Make phone and SIP calls”.
It is quite easy to install “Calls”. Just input following command:
sudo apt install gnome-calls
Please refer to the figure below for more details about its main window.
Gnome calls main window
Click menu “VoIP Accounts” to add a new SIP account. Most items are same with other SIP clients. For example, miniSIPServer is at “192.168.3.42” and the account is “100”. Please refer to the figure below.
Configure an account
Please pay attention that (1) the default port is 0, we need to change it to 5060; (2) we need to enable the account. Calls doesn’t display its status, so we can check miniSIPServer local users’ window to check their status.
When making outgoing calls, we can dial the target number in the “Dial Pad” panel directly.
Make a call out
If there is an incoming call, just answer or reject the call.
Incoming call
Obviously “Gnome Calls” is very simple at this time and it will be developed with more functions. If we are deploying a simple VoIP network and hope to build all elements on Linux, it could be a good choice.
Ubuntu 22.04 (Jammy Jellyfish) is the latest LTS (long tem support) version, so it is quite important for miniSIPServer to support this version.
We try to install and run miniSIPServer on Ubuntu 22.04, it is very easy! And we are very glad that it is perfect to run miniSIPServer on it. Please refer to the figure below.
miniSIPServer on Ubuntu 22.04
If you are going to deploy a new VoIP network, it could be a good choice to run miniSIPServer on Ubuntu 22.04.
Debian 11 (Bullseys) was released at August 14th, 2021. It is the latest stable version and very important for business deployment. So we run miniSIPServer on this system and make some tests. The result is perfect!
It is no problem to run miniSIPServer on Debian 11. You can refer to online document for more details about how to run miniSIPServer on Linux systems.
Today we add a new version (for 5 clients) into miniSIPServer lines. This version is FREE! That means you don’t need a license and don’t warry about expired problem.
This “5 clients” version is perfect for small VoIP delpoyment, such as family communication, testing and so on. You don’t need to pay a cent to get full VoIP functions. Of course, the clients are limited to be no more than 5 clients.
In another way, free versions are not available for commercial usage.
Sometimes, we want to know the details of real-time call status, such as who is calling, how many incoming calls, and so on. In fact, miniSIPServer has a real-time CDR window which can display CDR records just finished their calls. That means the real-time CDR window only has a half real-time function, and it cannot display the details of calls which are still in calling status.
Obviously, we think we need to upgrade such windows to display more details. And here we get:
Truly real-time CDR window
In the new window, the call which has been released or disconnected is in gray. If the call is calling, it will be in black. When the called party is ringing, it will be changed to be in blue. Once the called party answers the call, it will be red immediately.
So you can get real-time status of all concurrent calls. That’s interesting.
Of course, you need to configure miniSIPServer to display real-time CDR and what kinds of CDR should be generated. Please click menu “data / system / call detail record” and get configuration below.
CDR items
Most important, you need to configure “generate fail call CDR” and “show real-time CDR information” items. Please visit manual document for more details about other items.
When deploying a VoIP network, we often have one-way or no-way audio problems. It is caused by private network, for example, some SIP phones or miniSIPServer are behind routers and other SIP devices are in another different network which could be a private network or public network.
To resolve such problem, we often suggest to configure “forwarding ports” in routers manually. If you are familiar with routers, it is easy to do that.
But someone might not know how to do that, or someone might make mistake in router’s configuration. So we add a new feature in miniSIPServer to help that.
It is UPnP (Universal Plug and Play). UPnP can help miniSIPServer to map necessary ports automatically.
Firstly, you need confirm that your router can support UPnP and it has been enabled.
Then, you can click menu “Data – System” in miniSIPServer and enable the item “Enable UPnP to ask router to map ports”. Please refer to following figure.
UPnP configuration in miniSIPServer
By default, miniSIPServer will map SIP (over UDP) port and audio ports for relaying audio streams.
In another way, there is a limitation in routers. Most routers limit the number of UPnP ports, for example less than 30 ports. So if you are deploying a miniSIPServer for 50 clients or more, you will still have to configure “forwarding ports” manually.
Deepin is a very popular Linux distributor system in China market. It is very beautiful and easy to use. More and more software have been migrated to this system in China. As we know it is based on Debian system, we think it should be no problem to run miniSIPServer on it directly.
And it is true! Follow the online document, we can install and run miniSIPServer as same as what we do in Debian system.
miniSIPServer in Deepin market
Yes, this system is very beautiful. After install miniSIPServer, you can find it in its software market.
Debian 10 (Buster) is released. It is a stable and important version and can be deployed in business environment, so we must pay enough attention to this version.
We make some test with miniSIPServer on Debian 10. Now we are proud to announce that it is perfect to run miniSIPServer on the latest Debian system. Please refer to following figure.
miniSIPServer on Debian 10
You can update Debian source list, then download and install miniSIPServer. No more action!
Some customers report a crash problem to us. All of them deploy “SIP over TLS” in their VoIP networks. We have upgraded miniSIPServer to latest V35 (build 20190313) with following key modifications.
(1) In the latest miniSIPServer, SSL library has been upgraded to the latest version.
(2) Only TLSv1.2 method is kept, that means SSLv2, SSLv3, TLSv1 and TLSv1.1 are cut. When we did research on customers’ problems, we found some bad guys were trying to use the bug of SSLv3 to hack into MSS. We have to move all these methods out to defend that. In future, we will add other methods, such as TLSv1.3. At this time, we need confirm SIP phones can support TLSv1.2 too if we want to deploy SIP over TLS.
In another way, we refine “SIP over TLS” document to provide a simple demo on how to create certificate files.
