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.
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.
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.
We upgraded cloud miniSIPServer system for some key features. The most important feature is “SIP over TLS”.
By default, cloud system opens TCP port 6060 to accept “SIP over TLS” messages. It is used to encrypt SIP messages. This feature is available for all virtual servers without any additional fee or configurations.
Now, SIP phones can connect to cloud miniSIPServer nodes with “SIP over TLS”, but “external line” and “SIP trunk” still can only use “SIP over UDP” to work with voip providers.
This feature can only encrypt SIP messages. If you want to encrypt media streams, such as audio stream and video stream, you need enable SRTP in your SIP phones. By default, media streams are bypass and processed by SIP phones themselves, cloud miniSIPServer will not process these media streams.
Please visit online document “SIP over TLS” for more details.
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.
Our data center have scheduled maintenance windows during which the basic infrastructure will be updated.
During the maintenance window, all our Cloud-MSS nodes will be powered down and that means all virtual servers will stop working. While a two-hour window is allocated for the maintenance, the actual downtime should be much less.
The scheduled maintenance window is set for: Tuesday, January 8, 8:30 AM UTC
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
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.
In this document, we provide many interfaces almost cover all necessary interfaces for basic calls, such as SIP trunk, external lines, routing, and so on.
Hope it can be helpful for your solution. If you need us to provide more interfaces, please contact us. All suggestions are appreciated.
Sometime, we may fail to make outgoing calls through SIP trunk if its service provider has some problem, such as no enough resources, and so on. If customers configure several SIP trunks and they are from different service providers, we can configure miniSIPServer to try another SIP trunk to continue outgoing calls.
In the SIP trunk “outgoing call” configuration, please configure a chained SIP trunk described in below figure.
