In previous versions of miniSIPServer, in order to enable “SIP over TLS”, it was necessary to configure certificate and key files (including self-signed certificates and keys). If these files were not present in the configuration directory, miniSIPServer would not enable SIP over TLS by default.
Most customers deploy “SIP over TLS” using self-signed certificates and keys. Linux systems come with the openssl tool built-in, making it very easy and convenient to create these files. However, Windows systems do not have the openssl tool by default, requiring customers to download the tool to create certificates and keys, which is slightly more troublesome.
To reduce the workload for our customers, we have streamlined the steps for enabling “SIP over TLS” in miniSIPServer:
miniSIPServer now enables “SIP over TLS” by default. If certificate and key files are configured, it uses the customer’s provided certificates and keys to encrypt SIP messages. If no certificate or key files are configured, miniSIPServer automatically creates a self-signed certificate and key to encrypt SIP.
Therefore, when miniSIPServer starts, we can see the TLS port information, indicating that “SIP over TLS” has been enabled.
The “hunting group” is a long-established enterprise communication service that was widely used in the circuit-switched telephone era and remains deployed by many businesses even in the VoIP era. However, times have changed, and the service itself must evolve to keep pace with the characteristics and requirements of IP networks. Based on recent customer needs and changes in the network environment, we have implemented several optimizations to the hunting group feature in miniSIPServer.
The focus has been on modifying and optimizing the “Operator” feature within the service. Please refer to the image below:
Change 1: One operator can now be assigned to multiple hunting groups simultaneously. Previously, an operator was restricted to a single hunting group, which no longer meets the needs of modern enterprises. As employees often handle multifaceted roles, there is a significant need for them to support multiple hunting groups at the same time. This new feature addresses this requirement.
During the era of circuit-switched telephony, phone terminals lacked sufficient capabilities. Therefore, hunting groups typically allowed operators to log in or out by dialing specific codes. However, for the following reasons, the new hunting group feature in miniSIPServer no longer supports manual operator login or logout:
(1) Most modern SIP terminals now have sufficient functionality to implement features like “Do Not Disturb” directly on the device side, making manual login/logout unnecessary.
(2) Now that one operator can support multiple hunting groups simultaneously, simple login/logout actions are inadequate. Operations would need to be performed for specific hunting groups, making dialing procedures cumbersome and unnecessary.
Change 2: For hunting groups using the “Linear” policy, operators can now be assigned a sequence number to define their selection order. Previously, the selection order was based solely on the sequence in which operators logged into the system, which essentially resulted in a random order and could not meet practical requirements. In real-world scenarios, certain operators often need different priority levels within the group.
An operator with a smaller “Linear sequence number” will be selected earlier by the hunting group. If multiple operators have the same sequence number, they will be sorted by their login time, with those who logged in earlier receiving priority.
Of course, this new configuration does not apply to the “round-robin” strategy. The round-robin strategy always strives to distribute calls as evenly as possible among operators to balance the workload.
The hunting group feature has been updated in both the on-premises and cloud versions of miniSIPServer. There are no differences in configuration or usage between the two versions. Please refer to the product documentation for more detailed information.
Enterprise communication systems are typically deployed within private networks, with Session Border Controllers (SBCs) or voice gateways installed at the network edge to facilitate external communication. Therefore, in most scenarios, enterprise communications remain highly secure. However, a growing number of businesses are now deploying SIP servers in the cloud, while an increasing volume of SIP terminals within enterprises are accessing these corporate SIP servers from external networks. This shift has exposed part (or all) of enterprise communication systems to public networks, making security concerns increasingly severe.
The security of enterprise SIP communication involves many aspects of the network system, such as firewalls. Focusing solely on the SIP communication itself, it must be encrypted to prevent the exposure of communication information to other network users. Encrypted SIP communication consists of two parts: (1) SIP message (signaling) encryption, and (2) voice stream (RTP) encryption, as illustrated in the figure below:
Certainly, enterprises can deploy VPNs to encrypt the entire network system — not just communication systems but also office systems and more. Encrypted SIP communication can also be established over a VPN. However, setting up an enterprise VPN involves relatively high costs and complex systems. This article focuses solely on encrypted SIP communication and does not cover other network security technologies such as VPNs.
SIP message encryption is achieved through “SIP over TLS.” Both cloud-based miniSIPServer, on-premises miniSIPServer, and miniSIPPhone support SIP over TLSv1.2 / TLSv1.3. Please refer to the online documentation for details, as this article will not elaborate further on this topic.
Voice streams are encrypted through SRTP or DTLS-SRTP transmission. The master key and master salt for SRTP are transmitted and negotiated via the SDP (RFC4568) in SIP messages. Therefore, only when SIP messages are encrypted can the critical information of SRTP be ensured not to be leaked. Simply encrypting voice streams with SRTP while transmitting SIP messages in plaintext cannot guarantee the overall security of SIP communication.
RFC4568 defines several cryptographic suites. Currently, we have chosen to support the default AES_CM_128_HMAC_SHA1_80 and do not yet support other encryption suites.
The SRTP protocol family includes numerous extensions. Currently, miniSIPServer and miniSIPPhone support the most fundamental RFC3711 protocol, which is also the basic SRTP protocol supported by the vast majority of SIP devices (including servers, PBXs, SBCs, and endpoints). miniSIPServer can also support RFC5763 which is the basic protocol for DTLS-SRTP. (At present, some SIP clients don’t support DTLS-SRTP, so if you want to deploy that, please check their capabilities carefully.)
miniSIPServer and miniSIPPhone can enable SRTP by default without requiring additional configuration. Some SIP devices need explicit configuration to select SRTP. For example, when configuring an account in MicroSIP, the “Media Encryption” setting must be configured as follows:
Debian 13 (Trixie) was released yesterday. It is the latest stable version and quite suitable for business deployments. We are big fans of Debian, so we immediately run and test miniSIPServer on this system. All test cases are passed. Perfect!
You can deploy enterprise VoIP network with Trixie, it is an exciting choice.
As we known, the VoIP (SIP) domain always uses SIP URI to establish call sessions. To work with traditional PSTN networks, we need gateways (or SBC) to bridge two networks. Most of these gateways can support SIP URI, so we can always use SIP trunk to estanlish connections between VoIP and PSTN with SIP URIs which are as same as connections between VoIP domains.
But some gateways cannot support SIP URIs, they can only accept traditional telephone numbers which are the tel URIs defined in RFC3966. The URI is in “<tel: xxx>” format, not in “<sip:name@address>”format. Please refer to the figure below.
miniSIPServer can always accept the tel URI from peer sides, but never send out the tel URI. In recent months, several customers ask us to support sending out the tel URI through SIP trunks to work with some PSTN gateways. So we upgrade miniSIPServer to V60 (build 20250208) to update the SIP trunk functions. In the “outgoing call” of SIP trunk, we can select “Use the tel URI” item, then miniSIPServer will use <tel> URI to make outgoing calls for the SIP trunk.
For incoming calls of the SIP trunk, it is unnecessary to configure anything since miniSIPServer can accept both SIP URI and TEL URI.
For miniSIPServer V60, the lowest requirement of Debian is changed to the “oldoldstable” version which is Debian V10 at this time. That means Debian V8 and V9 will not be supported now.
Of course, the lowest requirement of Ubuntu is also changed to Ubuntu V18.04.
Please refer to the document for more details about Linux systems.
miniSIPServer is upgraded to V60 which is the latest stable version for business development. The first big thing is “conference room” feature which provides conference calls for local users. At most 5 clients can join the same conference call. Please refer to the service document for more details. Cloud miniSIPServer is also upgraded to support this feature.
In another way, as we have posted in previous blog, several services are finally removed from local miniSIPServer, such as calling-card and call-shop. These features were important for some of our customers, but it is time to say good-bye now.
As we know, miniSIPServer was developed about 20 years ago. Lots of services and features are added into miniSIPServer to support more and more customers.
Recently we have reviewed all these services. Some services have so long history that we have to think whether they are suitable for current environments, for example call-shop, calling card, and so on.
Next version will focus on refining or clearing some services. miniSIPServer will step into next stage and be more faster, more stabler.
Run miniSIPServer on Ubuntu 24.04 LTS (Noble Numbat)
Ubuntu 24.04 is the latest LTS (long-term support) version, so it will be deployed widely in business environment. We install miniSIPServer on this important version and make some tests. The result is perfect! Please refer to the figure below.
If you want to deploy a new VoIP network on Linux system, Ubuntu 24.04 could be a good choice.
Please refer to online document for more details about how to run miniSIPServer on Linux system.
miniSIPServer recently is upgraded to support TLSv1.3. This modification doesn’t affect configuration, so you need to do nothing if you upgrade your miniSIPServer to the latest versions.
Two modules could use TLS transport: (1) SIP server, and (2) Embeded HTTP server. If your SIP phones can support TLSv1.3, it is better to use TLSv1.3 to protect communication. Please refer to “SIP over TLS” document for more details. Both local miniSIPServer and cloud miniSIPServer can support SIP over TLSv1.3 now.
By default, miniSIPServer starts an embeded HTTP server for web management. If you want to manage it through the pubilc network, you have to enable TLS transport to protect HTTP information. In another way, most navigators, such as Chrome, Edge, Firefox and so on, can support TLSv1.3 now. Please refer to “web management” document to enable encrypted HTTP.
