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ARM64 and some modification

ARM64 and some modification

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++

181 “Call Is Being Forwarded”

181 “Call Is Being Forwarded”

“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.

Call fowarding with 181 messages

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"

RequestURI parameter of external lines

RequestURI parameter of external lines

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.

Additional parameter of Request URI configuration

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.

New web UI for miniSIPServer

New web UI for 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.

FREE version

FREE version

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.

Hope you can enjoy it!

Relay video streams

Relay video streams

With previous versions, if you want to configure miniSIPServer to relay media streams, miniSIPServer will only relay audio streams and discard video streams.

It is because video streams require much more bandwidth and calculation capability. Some devices cannot support that. But more and more customers require us to refine it to relay video streams at the same time since most devices are more powerful and they have enough bandwidth.

It seems reasonable and we think we need upgrade miniSIPServer to fit such requirements.

So the latest versions (build 20210604) are released. If miniSIPServer is trying to relay media streams, it will relay audio streams and video streams together.

You don’t need change your configuration. And please pay attention to your device capability and bandwidth.

An interesting feature

An interesting feature

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:

Real-time CDR window
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 configuration items
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.

Maximum concurrent calls of local user

Maximum concurrent calls of local user

Previous miniSIPServer versions only limit “maximum concurrent outgoing calls”, and didn’t limit the total concurrent calls. Normally, it can fit most requirements since we think SIP phones or SIP clients should be able to limit their incoming calls. In recent days, some customers response that their SIP phones don’t have enough functions and hope miniSIPServer to be able to limit total concurrent calls of each SIP phone. To fit this requirement, we upgrade miniSIPServer to V34. Please refer to following figure.

Maximum concurrent calls
Maximum concurrent calls

You can configure “maximum concurrent calls” to be zero. In this strange scenario, the SIP phone will never receive call and cannot make any outgoing call. It is to be noted that “maximum concurrent outgoing calls” should be smaller than “maximum concurrent calls” because “maximum concurrent calls” limits both outgoing calls and incoming calls together.

Trace on IP address

Trace on IP address

Previous miniSIPServer has a trace tool which is “trace all”. It can capture and trace all SIP calls which MSS receives or sends out. This tool is very useful when we build the VoIP network at the first step. But it is almost useless in an exist working environment.

It is dangerous to capture ALL SIP calls in a working system since there are too many SIP messages and inner information. By default, we can filter the call according to caller number or called number. In the recent V33 version, we disable “trace all” and replace it with “trace on IP address”. Please refer to following figure.

Trace on IP address
Trace on IP address

With this tool, we can capture a specific complete IP address, such as “10.0.0.101”. We can also set a part of IP address to capture some SIP calls from some IP addresses, such as “10.0.0”, in this scenario, all SIP calls from IP addresses begin with “10.0.0” will be captured. By the way, we can also set IPv6 address with this tool.

Now you can see this tool can be used in both lab environment and working environment.

miniSIPServer on Ubuntu 18.04

miniSIPServer on Ubuntu 18.04

It is perfect to run miniSIPServer on Ubuntu 18.04 which is the latest LTS version.

We have tested some scenarios with miniSIPServer on Ubuntu 18.04, everything is OK. We strongly suggest you to upgrade your Ubuntu system to this version.

miniSIPServer on Ubuntu 18.04
miniSIPServer on Ubuntu 18.04