ptime=20, 30, 40

ptime=20, 30, 40

In the SIP session, we can set ‘ptime’ parameter of SDP to negotiate RTP packages between two call parties. If one side has different idea about the RTP package size, it can send its own ‘ptime’ parameter back with the value it wishs. But the world is so complex, some SIP devices don’t care ‘ptime’ paramter and they don’t tell other sides which values they like, they just send RTP packages back directly. It could cause problem.

For example, as we know, ‘call recording‘ service requires miniSIPServer to mix two audio streams from both call parties. In order to make the work easy, miniSIPServer will set ‘ptime’ to 20 which means the call parties should send their audio streams in RTP packages every 20 milliseconds, then miniSIPServer get two packages every 20 milliseconds and mix them into local audio files. Now surprised, some devices send RTP packages every 30 milliseconds, some send RTP packages every 40 milliseconds. When miniSIPServer has to mix these packages, some bytes have to been lost since the time is different and the package size is different too. It makes the audio quality of local audio files very poor.

Of course, the perfect solution is all SIP devices respect ‘ptime’ parameter, but we cannot count on it.

So miniSIPServer is upgraded to V40 (build 20220922) to fix it. The new version will try to cache RTP packages of both sides and smooth the mixing procedures. And we have to say that it will increase the workload of miniSIPServer.

Say goodbye to Windows XP/2003

Say goodbye to Windows XP/2003

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.

For the 4K screen

For the 4K screen

Our customers report a bug when running miniSIPPhone with a 4K screen. Please refer to the figure below.

miniSIPPhone main window
miniSIPPhone main window mixed up

When building GUI (including miniSIPPhone and miniSIPServer), we use absolute positions and lengths to assign components, so they are quite small or tight if the screen has hight resolution ratio, for example, a 4K screen.

To fix that, we change the absolute values to relative positions and lengths. Of course, both miniSIPServer (V39 build 20220823) and miniSIPPhone (V8.4) are updated.

In another way, we refine dialogs sizes and styles of miniSIPServer at the same time. It should be more comfortable now.

Gnome Calls

Gnome Calls

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
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 VoIP account in Calls
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 outgoing calls
Make a call out

If there is an incoming call, just answer or reject the call.

Incoming calls
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.

Hope you can enjoy it.

miniSIPServer on Ubuntu 22.04

miniSIPServer on Ubuntu 22.04

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
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.

call tag and cause

call tag and cause

Sometimes, customers want to know some details about history calls, such as who released such calls and why they were released. Because they were not real-time information, we cannot use miniSIPServer trace tool to get the results, so we update CDR function to save more informations for deep research.

In the FCI (Furnish Call/Charge Information) section, we add two parameters “callTag=” and “cause=”. Please refer to the CDR records as below.

FCI parameters
FCI parameters

“callTag=” parameter will be used to save the position where the call is released in miniSIPServer. We can know where and who release the call. For example, the call might be released because it received a BYE message from caller side, and so on. The tag value is an inner value and it will not be published to customers.

“cause=” parameter is used to save the cause value. If miniSIPServer receives 4xx or 5xx messages from the called side, and there is “Reason” header which has ’cause’ parameter, miniSIPServer will use this value to release the call,otherwise, miniSIPServer will use inner cause value. And of course, this parameter will be saved in FCI of CDR records.

call-back service is updated

call-back service is updated

By default, miniSIPServer opens UDP port 5080 to receive call-back message to invoke calls. If miniSIPServer is deployed in public network, it is possible to receive lots of other UDP packages. Of course, we can configure “application server address” for IP address authorization, but unfortunately its default value is blank and it could be dangerous.

So we update call-back service to protect the system and refine its service logic. Please refer to the configuration window firstly.

Call back service configuration
Configuration

(1) The default value of “Application server address” is changed to a local loop address “127.0.0.1”, so outsides UDP packages will be disabled. Of course, we can still keep it to be blank to accept all packages from any address, but we suggest not to do that.

(2) Local listen port can be zero which is used to close UDP socket. If the port is zero, miniSIPServer will disable the whole call-back service since it is impossible to receive any outsides UDP packages now. If you don’t use call-back service, it is better to set it to be zero.

(3) Cancel “external line mode” item. Some cutomers always ask us what it is and always confused with this item. It is just to add ‘out group prefix” automatically before the called numbers in two sessions. In fact, it is not flexible if we want to call local users and outsides users at the same time. So we discard it, if we want to call outsides users, we can add ‘out group prefix’ manually in the REQUEST messages. That means application server should be responsible for the numbers format and dial plan result.

Please refer to call-back service document for more details.

start miniSIPServer on boot (Linux)

start miniSIPServer on boot (Linux)

With old Debian version, we can start miniSIPServer on boot by updating ‘/etc/rc.local’ file. Since most Debian versions have been migrated to systemd, so we have to change to other ways to do that.

The first way is to keep “rc.local”, so we need to enable “rc-local.service” in systemd. This service is disabled by default, we can use command below to activate it.

sudo systemctl enable rc-local

It is not a very good way because we have to update “rc-local.service” to fit miniSIPServer requirements which might affect other applications defined in the same “/etc/rc.local” file.

It is better to define an independent service to start miniSIPServer in systemd. In fact, it is quite easy. Lets explain the details in Raspberry Pi system for example. In this scenario, we will indicate Pi to start miniSIPServer (command line) at boot and the default user is “pi”.

First, please create a “minisipserver.service” file in directory “/lib/systemd/system” with content below.

[Unit]
Description=miniSIPServer
After=network.target mariadb.service
Requires=network.target mariadb.service

[Service]
Type=simple
GuesMainPID=no
User=pi
KillMode=process
ExecStart=/opt/sipserver/minisipserver-cli

[Install]
WantedBy=multi-user.target

Then, you can use command to activate this service.

sudo systemctl enable minisipserver

Once “minisipserver.service” is enabled, miniSIPServer (command line) will start up automatically when the system is reboot.

In the “minisipserver.service” file, we can see two important sections, [Unit] and [Service], and we need to describe more details about them.

Unit

Please click here to get more details about unit section in systemd.

We care “After=” and “Requires=” parameters. Since miniSIPServer is a network application, so it requires network is ready and need to be started up after the network. In our environment, miniSIPServer is connected with Mariadb/MySQL database, so it requires the database is ready too. If you don’t need database, you can remove “mariadb.service” from “After=” and “Requires=” parameters .

Service

Please click here to get more details about service section in systemd.

We care “User=” and “ExecStart=” parameters. “User=” parameter indicate systemd who is trying to run this service and start specific application. Since we are using Raspberry Pi system and the default user is “pi”, we set this parameter to “pi”. You need to change it to your own user name in your own Linux system. By default, miniSIPServer is installed at “/opt/sipserver” directory and the command line file is “minisipserver-cli”, so we set “ExecStart=” parameter to indicate systemd where to get the executing file and run it.

New CDR format

New CDR format

miniSIPServer old versions save CDR records into some binary files in “AppData/cdr” directory. If you want to check history records, you have to start miniCDR tool to open them. If you want to do more research, you will have to use miniCDR to convert them into CSV format files.

Now miniSIPServer new version ( I mean V39 or newer) will save CDR records into CSV files directly. Of course, all these .csv files are still stored in “AppData/cdr” directory. Since .csv files are text format, you can use any text tools, such as notepad in Windows and Gedit in Linux, to open them. If you have Excel, you can use it to open and analyze these CDR files directly too.

And miniCDR is upgraded at the same time to be able to process previous .cdr files and new .csv files.

Optimization: load IVR-XML files into memory

Optimization: load IVR-XML files into memory

In most scenarios, the size of IVR-XML files are very small, only several KB, so it is no problem to read these files from hard disk when processing an incoming call to trigger IVR service. But if there are heavy workloads, for example, huge concurrent calls to trigger IVR services, miniSIPServer has to read these files from hard disk frequently. Obviously, it will affect server’s performance.

So we try to optimize this and load all IVR-XML files into memory. If the file is not modified, IVR service will read it from memory directly. If the file is modified, miniSIPServer will reload it into memory automatically.

That means all IVR operations will be processed in memory and will never visit hard disk if the files are not modified. With this optimization, miniSIPServer will be faster than before.