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Category: miniSipServer ( local )

SIP server for windows and Ubuntu system

New LTS V32

New LTS V32

Finally, we release the latest LTS version V32. It has been a long time since we released the first V32 for test. During these months, we have updated web UI, GUI, SIP core and lots of items. This is a very exciting version and we will provide FIVE years tech support for it.

In another way, the latest stable version is V33 now. From this version, Linux (X86-32) systems will not be supported. New features and services will be developed on this version.

Hope you can enjoy these versions!

External lines configurations

External lines configurations

We often configure miniSIPServer to connect VoIP carriers’ network with external lines. There are lots of VoIP carriers and someone always asks us how to configure external line.

In our step by step document, we give a demo to configure MSS to work with “call centric”. You can refer to this document for more details about VoIP networks and external lines. In another way, we give some more examples in chapter “External lines” of F.A.Q document. Please refer to these documents if you are interesting in it and hope they can be helpful to you.

https://www.myvoipapp.com/docs/faq/index.html

 

miniSIPServer for Raspbian (Stretch)

miniSIPServer for Raspbian (Stretch)

We rebuilt miniSIPServer V32 for the latest Raspbian.

That means only Raspbian (Stretch) will be supported now. If  you are still running MSS on Jessie or older systems, please upgrade Pi before you prepare to upgrade MSS.

If you are still working on Pi1 or Pi2 and don’t want to upgrade Pi, you can still try MSS previous versions. For example:

V31 https://www.myvoipapp.com/download/backup/mss_v31/pi/mss_v31_pi_u20.deb

V30 https://www.myvoipapp.com/download/backup/mss_v30/pi/mss_v30_pi_u20.deb

Relay media streams of SIP trunk outgoing calls

Relay media streams of SIP trunk outgoing calls

In some VoIP scenarios, we need configure “SIP trunk” to work with VoIP providers or gateways. When processing media streams, we hope (1) local users/phones should process their streams by themselves without MSS, and (2) MSS should help to relay media stream for all outgoing calls to peer SIP servers or gateways.

To fit these requirements, we update MSS V32 to be able to configure “relay media” item in “SIP trunk”. Please refer to following figure for more details.

Configure "relay media stream" item in SIP trunk outgoing call
Configure “relay media stream” item in SIP trunk outgoing call

By the way, MSS can only relay audio streams at this time, so video streams will be lost if you want to MSS to relay streams.

Store your own audio files

Store your own audio files

miniSIPServer can support customers’ own audio files to replace default files. With previous MSS, customers have to backup and restore these files once they want to upgrade MSS.

This is a little trouble. With the new V32, we can resolve it now.

When MSS starts up, it will create a sub directory ‘cust_ann’ in ‘mss_ann’ directory, now all your own audio files can be stored in this directory. When MSS is uninstalled or upgraded, this directory and its files will not be deleted or replaced by default files, and MSS can get audio files from this directory directly when it starts up.

In windows system, it could be “d:/myvoipapp/minisipserver/mss_ann/cust_ann” directory by default. In Linux system, it could be “/opt/sipserver/mss_ann/cust_ann/”.

Please refer to our online document for more details about how to record own audio files.

https://www.myvoipapp.com/docs/others/how_to_record_your_own_audio/index.html

Connect to sonetel

Connect to sonetel

“sonetel.com” is a VoIP carrier who can provide local phone numbers. We can add “external line” to work with their servers. According to its description, there are some special items need to be cared:

  • Sonetel uses email address as SIP account and
  • It deploys SBC or proxy to process all incoming SIP messages.

Here we give a simplel example to describe how to work with Sonetel. We assume the SIP account is “abc@gmail.com”.

In MSS, please cilck menu “data > external line” to add a record.

Configure Sonetel lline
Configure Sonetel lline

In “Basic” tab, the line type should be “Connect to peer VoIP server”, the Account should be “abc” and the Domain is “gmail.com”.

By the way, the Password is the password you sign up in Sonetel, not your own email password.

Since all SIP messages are processed by sonetel SBC/Proxy, we need configure “outbound” information in the “Outgoing call” tab. Please refer to following figure.

Sonetel SBC
Sonetel SBC

The Sonetel proxy address is “sip.sonetel.com” which should be described in the email sent by sonetel.

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.

Work with Chinese CTC IMS network

Work with Chinese CTC IMS network

Yesterday, we helped a Chinese customer to deploy MSS to work with CTC IMS network. In this scenario, CTC IMS network has ZTE soft-switch (according to User-Agent header in SIP messages) , we need be careful to cooperate with it.

Since CTC provides user name and password for authorization, we configure “external line” in MSS to do that. Following sections will illustrate some key points.

Authorization user name

By default, we often use “External line (account)” as authorization user name, but ZTE softswitch requires full URI format, so we need configure “The authorization ID should include address information” in external line. Please refer to following figure for more details.

Authorization user name
Authorization user name

For example, if this item is selected, the authorization name will be “+8612345678@gd.ctcims.cn” according to above figure.

If it is not full format, IMS network will return “403 Forbidden” messages to reject it. In fact, we think it is a bug in ZTE softswitch since there is “realm” and “domain” parameters in SIP authorization header. No matter the user name is full format or not, the device should pass it according to successful authorization itself.

Anyway, if you have same problem to cooperate with other IMS networks, please pay attention to it and configure such item to take a try.

Proxy

In Chinese CTC-IMS network, its “SIP server” is logic domain, not a real SIP device and cannot be visited. In above scenario, “gd.ctcims.cn” is its domain, not its real address. SIP messages should be routed to another device (we think it is a SBC or proxy), so we need configure “Via” address in MSS external line configuration. Please refer to following figure.

SIP proxy in IMS
SIP proxy in IMS