We are very exciting to release miniSipServer new version 11! This is a great version and can support ‘Gtalk to SIP’ feature now.
What that means? It means you can connect your VoIP network with Gtalk network now. Gtalk users can call to your extensions directly, use your VoIP services, and so on. We bring Gtalk into your enterprise communication system. Funny!
Please refer to our online document for more details:
Yesterday we got and installed latest Xubuntu/Kubuntu/Ubuntu 12.10, then we installed latest miniSipServer V10.5 on these platforms.
The result is perfect! It is no problem to run MSS without any modification.
In another way, we would like to suggest Xubuntu and Kubuntu. According to our test result, Ubuntu system is very slow, however Xubuntu and Kubuntu are perfect.
As we know, most SIP devices use SIP over UDP by default, however, some SIP servers or communication servers only support SIP over TCP. We have to say that’s really strange. Our customers need work with such devices and we upgrade MSS to V10.5 to support SIP over TCP to fit this requirement.
It is very easy to configure MSS to support this feature. If you are interesting in it, please refer to following document for more details about it:
According to our customers’ requirement, we add a new product, 500 clients MSS, to our list. This version can support 500 extensions.
At the same time, we remove 1000 clients product from our public list which means you can still ask this product by sending mail to us. 1000 clients version is quite different with other versions, and it requires more help and support when customers try to deploy it, so we think it is not suitable for small business or SOHO business.
In another way, since Ubuntu LTS has been upgraded to 12.04 version, we upgrade all our work and test environments together, so MSS for linux version is upgraded to fit Ubuntu 12.04. We will not support other Ubuntu versions or linux distributions.
Some customers often request to deploy MSS behind NAT, but still need provide public service. That means MSS is in private network and some SIP phones/clients are in public network (internet), or MSS need work with public VoIP carriers’ network.
Following figure describes a simple network for this scenario:
In this network, we can see
(1) Private network is connected to public network by a router whose public address is 8.8.8.8 and private address is 192.168.1.1
(2) MSS is deployed in private network with private address 192.168.1.2.
(3) Some SIP phones are in the same private network, such as local users 100 and 101. Some SIP phones are in public network, such as local user 102.
It is no problem for local user 100 and 101 to visit MSS since they are in the same network. So the problem is how to make outside local user (102) can visit MSS.
We can resolve it by forwarding some ports in router.
First, in the router, we can configure forwarding UDP port 5060, 10000~20000 to the PC where MSS is installed. Most routers can support this function. Port 5060 is standard SIP port. Ports 10000~20000 are RTP ports to transfer media streams.
Second, we must indicate MSS to work with public address. Please click menu “Data / System / SIP” and fill the “Main address” with the public address “8.8.8.8”. SIP phones/clients can use this public address to visit MSS.
There is another problem. In above scenario, the router is configured with a fixed public address. In normal, the router could be ADSL router and it maybe has a dynamic IP address. Outside users cannot use the dynamic address to visit MSS. Then, how can we provide public services?
To resolve it, we can use domain name, for example, we can use DynDNS to provide domain name for our MSS. The router must be able to support “Dynamic DNS”. In our example, we assume we get a domain name “sip.dyndns.org” from DynDNS and configure it in our router, then we can use this domain name as another miniSIPServer address. In following figure, we use such domain name as main address, and use the private address as additional address.
SIP phones/clients must be able to use domain name as server address or proxy address, so they can configure “sip.dyndns.org” to visit MSS in our scenario and make calls.
Local miniSipServer (stable) is upgraded to V6.1.4.
In this version, trial duration is changed from 20 days to 30 days. Some customers need more time to test MSS full features. We think it is very reasonable, so V6.1.4 or higher can support more trial duration.
In another way, this version is updated to support “direction” parameter in PRESENCE NOTIFY message which is very necessary for call center applications. This parameter is used to indicate the operator current call is outgoing or incoming.
Gianfranco, one of our partners, released a new softphone SDK. This SDK provides full SIP features, such as HOLD, Transfer, and so on. It works as .net component or .com component. It is easy to build a new softphone base on this SDK.
Most important, it has been tested with lots of SIP servers or proxy, such as our SIP server (miniSipServer), Asterisk, OpenSER and so on.
If you are interesting in building your own SIP softphone, this SDK could be a very good choice for you.
miniSipServer V6 is upgraded to V6.1.3. This version can support SIP INFO message to transport DTMF signals.
Some VoIP carriers’ servers or VoIP gateway can only support this INFO message to transport DTMF signals which are very necessary for IVR services, such as auto-attendant.
There are several content type in INFO message to transfer DTMF. MSS can support “application/dtmf-relay” content type.
Now, MSS can support INFO and RFC2833 for DTMF signals.