SIP server for windows and Ubuntu system
webRTC is defined in HTML5 to support real-time communication. We spent several months in this technology and release the latest V18 to support it finally!
Now you can connect Chrome to miniSIPServer to work as local user. It is so interesting and hope it can help you to deploy your own VoIP network.
Please refer to following document for more details about webRTC in MSS:
http://www.myvoipapp.com/docs/mss_services/webrtc/index.html
Ubuntu 14.10 is released today. We download and install it in our lab and make some test with miniSIPServer V17 (for linux).
It is no problem to run miniSIPServer on the latest Ubuntu system. Please enjoy it!
V16 has been released for several days and response messages from our customers are very exciting, so we decide to upgrade LTS version to V16 since it can support very important feature: IPv6.
At the same time, the latest stable version is upgraded to V17. In this version, something is changed:
“Event channel” is based on websocket, and customers’ applications can establish websocket connection to get MSS inner call status. Please refer to online document:
http://www.myvoipapp.com/docs/mss_services/event_channel/index.html
“MSS trunk” is our private protocol to bypass some ISP blocking. It helps some customers to deploy VoIP network at their sides. At this time, since MSS can support SIP over TLS, we think this feature could be replaced by it, so we decide to remove this feature from V17 and abover versions.
We hope new versions can fit your requirements and benefit your VOIP network. Please enjoy them and update us if you have any suggestion or question.
As we know, IPv6 is next internet protocol. All SIP devices will be migrated to IPv6 since IPv4 network has more and more problems.
Today we release miniSIPServer V16. The big feature of this version is to support IPv6. You can find it is so eay to deploy a voip network over IPv6 with miniSIPServer.
Please refer to following document for more details and enjoy IPv6 now!
http://www.myvoipapp.com/docs/mss_services/ipv6/index.html
And V16 will be next LTS version once we think it is ready for that.
csipsimple is a very good sip client software in Android, we often suggest our customers to use it if they want to deploy voip network with Android phones.
As we know, there are always one-way or no-way audio problem. To solve this problem, STUN should be configured in softphone. But some customers often response it is hard to set STUN in csipsimple.
In fact, it is easy to do that. Please follow below steps which are described in csipsimple website too:
(1) Go on setting Settings > Network – Tick “Use Stun” and fill a stun server on the field bellow. If you are cloud-mss subscriber, you can use your virtual server address as your STUN server too. Of course, you can use csipsimple default STUN server.
(2) You can also try to use ICE in addition to STUN if STUN alone doesn’t solve the problem : Settings > Network – Tick “Use ICE”.
Today we download the latest Ubuntu 14.04 and install it in our lab to make some test. It is no problem to run miniSIPServer on this system. And it seem V14.04 is better than its previous version.
So if you are interesting in Linux or Ubuntu, you can try this version.
Today we release latest V15 for miniSIPServer. This version is focus on providing a new service engine which is written in Python script.
That means almost all services are written in Python script files. New service engine is more flexible to fit different services requirements. Some advanced customers even can written their own special services now.
Today we upgrade miniSIPServer V14 to support 64 bits Debian and Ubuntu. That’s a great news for some of our customers.
As we know, in some networks, customers have deployed 64 bits Debian or Ubuntu servers. So when we want to install previous 32 bits miniSIPServer in their servers, it is required to install some additional 32 bits libraries. Of course, it is no problem to do that, but some customers often think it could be unstable for their servers.
So we decide to migrate MSS to 64 bits to fit this requirement. Now it is unnecessary to care about 32 bits problem.
For customers who are using 64 bits Windows, it is unnecessary to care about 32 bits problem. Windows can process them in perfect way, so please keep current versions.
V14.4 is updated to support a new feature in “dial plan” process. This feature is “refine called number”.
“Refine called number” can be used to refine called number before calls are routed to external lines or SIP trunks. It is the last chance to change called number to fit different requirements from peer VoIP servers.
For example, one of our customer has two VOIP accounts. One is from local provider, another is from international provider. These two VoIP providers have different number format requirements, and our customer only want to has one kind of dial plan for both of them. So we can configure “refine called number” to refine the final destination number to fit it. This scenario is illustrated below.
As described above, there are two VoIP accounts, and users need dial “90xxxx” to make outbound calls. “9” is MSS default outgoing call prefix. “0” is required by local VoIP provider. At the same time, the international VoIP provider requires that the number format should be “0086xxxx”.
After compare these number formats, we can find that we only need change prefix “0” to “0086” for international VoIP account.
Please click menu “Data / External line” and select the account to edit, then please click “Outgoing call” tab and configure following item:
Outgoing group ID = 1
We need configure a new record to change preifx “0” to “0086”. Please click menu “Dial plan / Transition” to add a new record:
Transition ID = 1 Transition type = Replace Start position = 0 Length = 1 Replace string = 0086
Please click menu “Dial plan / Refine called number” to add a new record:
Outgoing group ID = 1 <== defined in step 1 Called number prefix = 0 Transition ID = 1 <== defined in step 2
Here we maybe have a problem: the called number prefix is “0”, why? why not analyze “9” prefix? It is because that “9” has been deleted in “analyze called number” procedure and the number has been changed to “0xxxx” before it is sent to external line or SIP trunk, so we should analyze prefix “0” to refine final called number.
One of our customers has two different VoIP accounts, for example (1) 1234 and (2) 5678. It is required to select account “1234” if users dial “9xxxx” numbers and select account “5678” if users dial “8xxxx” numbers. The final numbers should delete these prefix “9” or “8” and “xxxx” should be sent to VoIP providers.
We can use MSS powerful “dial plan” features to fit this requirement.
By default, MSS uses called number prefix “9” to distinguish outgoing calls to outsides. If there are several external lines and without any special configuration, MSS will select one of them in round-robin for each call. Now what we need do is to configure different called number prefix and select different external line for them.
In this step, we need configure a record to delete number prefix “8” or “9” from called numbers. Please click menu “Dial plan / Transition” to add a record illustrated below.
Transition ID = 1 Transition type = delete Start position = 0 Length = 1
According to requirement, we need indicate MSS to analyze called number prefix “8” and “9” to use different specific external line. Please click menu “Dial plan / Analyze called number” to add two records.
Record 1: analyze called number prefix “9”
Dial plan = default Called number prefix = 9 Route type = external line Specific external line = 1234 <== use specific external line Change called number = yes Transition ID = 1 <== configured in step 1 Re-analyze after transition = no
Record 2: analyze called number prefix “8”
Dial plan = default Called number prefix = 8 Route type = external line Specific external line = 5678 <== use specific external line Change called number = yes Transition ID = 1 <== configured in step 1 Re-analyze after transition = no