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Category: Releases

Product release messages, including LTS, stable, development versions and patches.

Customized resource files

Customized resource files

Sometimes we need to use customized resource files, such as audio files, special IVR flows, certificate files and so on. With previous versions, all these files are stored in miniSIPServer install directory or its sub directories.

It could be a management problem when trying to uninstall or upgrade miniSIPServer. We have to be very careful to backup our own resource files.

V38 (build 20210108) is released to refine this. Customized resource files are separated to be stored in independent directories which are in application data directory.

For example, customized audio files will be stored in ‘custAnn’ directory. Once miniSIPServer is uninstalled or upgraded, these audio files will be kept without any affection.

Please refer to online document for more details.

A small thing: UPnP

A small thing: UPnP

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
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.

Monitor events in IVR call flow

Monitor events in IVR call flow

In miniSIPServer, we can use IVR-XML script to enable our own services, such as automatic-attendant. With previous IVR-XML set, ‘callto’ action will invoke a call to destination and finish the whole IVR process.

But if we want to monitor some events in the call flow, such as we want to check ‘busy’ event and change the IVR flow to a new action, what should we do?

Now V37 is released and a key feature is updated in IVR-XML. We can use ‘monitor-events’ in ‘callto’ action to monitor some events and change the call flow if they are caused.

For example, the ‘callto’ action can be configured as below.

<action method="callto" name="mainAction">
    <destination>100<destination>
    <monitor-events>
        <monitor-event detection="busy" nextaction="callto101"/>
    </monitor-events>
</action>

In this example, if the call invoked by ‘callto’ action is busy, IVR procedure will be changed to next action ‘callto101’.

Please refer to IVR-XML document for more details about “monitor-events” element.

IVR-XML exampleDownload

Above zip file is an example of new ‘callto’ action. You can save and unzip it into ‘xml’ sub-directory where miniSIPServer is installed and configure a new record to test it.

Configure miniSIPServer to trigger IVR-XML
Configure miniSIPServer to trigger IVR-XML
Refine “SIP over TLS”

Refine “SIP over TLS”

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.

Public address of SIP server

Public address of SIP server

With the latest miniSIPServer version, it can be configured with several addresses, such as “local address” and “public address”. Please refer to following figure.

local address and public address

In normal, if miniSIPServer is deployed in public network with public address, it is unnecessary to configure independent “Public address” in above SIP configuration since the local address is public address itself. But in some network environment, for example, miniSIPServer is in a NAT and need serve outside users, we need configure “public address”. Outside users can use this “public address” to work with miniSIPServer.

If the network bridges several sub-networks, such as private network and public network, several different private networks, and so on, some SIP phones MUST use miniSIPServer private address, others use miniSIPServer public address, here we suggest to enable “relay media” in local users’ configuration which will make miniSIPServer to relay audio streams between these networks.

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

New web

New web

We have updated our website with the latest bootstrap v4. Please take a look at it.

https://www.myvoipapp.com/

The most important is that you can visit it by using PC, mobile and tablet.  Hope you can enjoy it!

In another way, we are tied to remove spam posts in forums, so we decide to say good bye to it. If you have any suggests or problems, please feel free to contact us directly.

https://www.myvoipapp.com/contact.html

V32 (stable) is pre-released

V32 (stable) is pre-released

V32 (stable) has been tested in most scenarios and we are proud to release it today!

As you can see, this V32 is in stable branch. When we finish all test scenarios and get enough response messages from customers, it will be upgraded to LTS branch and the new LTS will be released finally in the begin of next year.

Please download it from our website directly.

https://www.myvoipapp.com/download/

Hope you can enjoy it!

Say goodbye to V24

Say goodbye to V24

It has been two years since the first V24 was released. It is the second LTS version of MSS. Now it is time to say goodbye. The latest LTS version will be V32 and we will provide five years support for it.

V32 will base on current stable version which is V31. We hope to do enough test on V31 as much as we can, so we decide to remove V24 linker from download page and only keep V31 linker. According to our test and customers’ experiences, V31 is very stable now. It will be a good choice to install or upgrade previous MSS to this version.

V32 is on the way and will be released in the beginning of 2018.

Final V31

Final V31

We released the final V31, that means we will be focus on next very important version V32 which will be our next LTS version to replace V24.

In fact, lots of features have been merged into the latest V31, and we will stay with V31 for several months since it is the base line of V32.  Please refer to following sections for more details about the key points.

Tools upgraded

In Windows platforms, we upgrade several important tools for V31.  The VC++ is upgraded to VC2010, so new MSS is using VC2010 run-time libraries. It could be powerful and better than previous VC2008 which has several manifest problems in customers’ environments.

The basic SSL library is migrated from OpenSSL to LibreSSL in MSS for windows. In Linux system, we still keep OpenSSL at this time and will move to LibreSSL in future. LibreSSL provides official windows library and we think it is optimized to be better than OpenSSL. If you are deploying “SIP over TLS”, this modification could be much better and safer then previous versions.

SIP stack upgraded

In recent days, we work with several customers to process scenarios with different IMS networks. We have to say we met several strange and very old SIP call flows. That’s ok, V31 is refined to fit these requirements.

“18X with/without SDP” flows are supported. “18X” means 180 or 183, so you can see several possibilities, such as “180 with SDP”, “180 without SDP”, “183 with SDP”, “183 without SDP”, and so on, and their orders are different. Sometimes we receive 180 firstly, sometimes we receive 183 firstly. In most scenarios, these messages are used to play different ring-back tone, so it is not only something with SIP stack but also something with media connections which means MSS inner MG module is upgraded too.

Another key point is SIP-UPDATE. Some IMS networks don’t use 18x to bring ring-back tone media information, they use SIP-UPDATE messages.  In another IMS network, we find it use “SIP-UPDATE without SDP” to keep alive in dialog. It is an interesting topic and we hope to write another blog to describe these scenarios carefully. Anyway, V31 is upgraded to support part of SIP-UPDATE to work with such IMS networks. We don’t implement all features about SIP-UPDATE and MSS will not invoke SIP-UPDATE flow by itself. If MSS wants to change media, it always invokes reINVITE procedures.

“tel” number format is supported in V31. Traditional soft-switch networks could transfer this format to MSS when they work with PSTN networks. We don’t understand why these soft-switch don’t convert it to SIP URL. Now V31 can accept that. Of course, MSS will never send out such number format.