Browsed by
Tag: sip

miniSIPPhone V26.1

miniSIPPhone V26.1

The latest version of miniSIPPhone V26.1 has been released recently, which primarily includes the following key features or modifications:

1. support DTLS-SRTP

After miniSIPServer added support for DTLS-SRTP, we updated miniSIPPhone to enable encrypted voice stream transmission via DTLS-SRTP. When deploying enterprise communication networks, especially those involving external public cloud systems, we fully implement high-strength encryption for both signaling and media to ensure the security of enterprise communications.

In both miniSIPServer and miniSIPPhone, we have uniformly implemented the following restrictions for DTLS-SRTP:

(1) DTLS must be DTLSv1.2 or above.

(2) The encryption suite is fixed to SRTP_AES128_CM_SHA1_80. Although the specification defines several encryption suites, we use the highest-strength encryption and do not support negotiating other encryption suites.

(3) The fingerprint always uses SHA-256 encoding and does not support SHA-1 or other encoding methods.

2. Simplify SIP account configuration

In the new version, when configuring SIP accounts, there is no longer a need for separate configuration to specify the port, as shown in the figure below:

Typically, SIP servers use standard ports to provide services, and users do not need to understand the port information specified by the protocol (just as we rarely specify or know about ports like 80 and 443 when accessing the internet). Therefore, we have removed the “Server Port” configuration option.

However, there are cases where SIP servers use non-standard ports (for example, miniSIPServer Cloud uses port 6060 for SIP-TLS access instead of the standard 5061 port). In the new version, we can specify both the address and port information together in the “Server Address” field, for example:

15000.s2.minisipserver.com:6060

If the server provides an IPv6 address and a non-standard port, we can configure it using the following example method:

[fe80::5a11:22ff:fe74:8198]:6060
Optimize “SIP over TLS”

Optimize “SIP over TLS”

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.

Secure enterprise SIP communication

Secure enterprise SIP communication

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:

Secure enterprise SIP communication network topology

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 carefullu check their capabilities.)

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:

MicroSIP SRTP configuration
Welcome! Debian 13 (Trixie)!

Welcome! Debian 13 (Trixie)!

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!

Run miniSIPServer on Debian 13.

You can deploy enterprise VoIP network with Trixie, it is an exciting choice.

miniSIPPhone supports SIP over TCP/TLS

miniSIPPhone supports SIP over TCP/TLS

Yes, we upgrade miniSIPPhone. Again!

miniSIPPhone V10.10 can support SIP over TCP and TLS now. In the account configuration, there is a new item ‘Transport’ to indicate which transport should be used to connect to SIP server.

miniSIPPhone account configuration, including transport configuration.

If SIP is over TLS, the messages are encrypted. It is quite necessary for enterprise communication if the servers or clients are deployed in public networks. As we know cloud miniSIPServer can support SIP over TLS and all virtual servers are deployed in the public network, so if you deploy miniSIPPhone at the same time, it could be safer for the whole VoIP network.

Of course, miniSIPPhone can work with other SIP servers who can support SIP over TCP/TLS to build a complete and safe enterprise VoIP system.

Send or receive instant messages

Send or receive instant messages

The latest version of miniSIPPhone is released today to support two key features: (1) Contact, and (2) Instant messages.

It has a new window to create and manage contact list like belowing:

miniSIPServer contact list

In the contact window, you can select the target user and double click it to make a call out, or you can press ‘C’ key or click ‘Call’ button to do that.

If you want to send instant messages, you can select the target user and press ‘M’ key or click the ‘Message’ button, then you will get instant messages’ windows:

Instant message window on Windows system
Instant message window on Linux system

One instant message window is used for one user. Each window has three areas: (1) Display area. It displays both incoming messages and outgoing messages. (2) Input area. You can input the instant message content here, and press ‘Ctrl+Enter’ keys to send the message out. (3) ‘Send’ button. Click it to send the message out.

At this time, miniSIPPhone uses SIP-MESSAGE to send and receive instant messages, and can only support plain text messages, so you cannot insert images, files, audios and videos into the messages.

Of course, miniSIPPhone can run on Windows system and Linux system (including AMD64 and ARM64). In fact, the users in above figure run miniSIPPhone on different systems.

Hope you can enjoy it. 🙂

miniSIPPhone for Linux (Debian/Ubuntu)

miniSIPPhone for Linux (Debian/Ubuntu)

Finally, miniSIPPhone is upgraded to V10. The most important thing is that it can support Linux system now. Of course, the distro must be Debian or Ubuntu. As same as miniSIPServer, Debian must be V10 (Buster) or higher versions, and Ubuntu must be V18.04 (Bionic Beaver) or higher versions.

Both X86_64 (amd64) and ARM64 (AArch64) are supported.

It is quite easy to run SIP phone on Linux system now. Please visit our website to download the latest version:

For example, you download “msp_v10_amd64.deb” and install it with following command:

sudo dpkg --install msp_v10_amd64.deb

Then you can click the linker to run miniSIPPhone:

If you want to uninstall miniSIPPhone, you can run following command directly to remove it:

sudo apt remove minisipphone
Conference room and others

Conference room and others

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.

Refine miniSIPServer

Refine miniSIPServer

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)

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.

Run miniSIPServer on Ubuntu 24.04

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.