The latest version of Sipwise’s open-source platform is now available: the Sip:provider v2.6!
I am excited to announce the general availability of Sip:provider v2.6, CE and PRO!
This release marks a big milestone because it’s the first open source turn-key platform with full presence support to provide Skype-like services.
What is the Sip:provider platform?
The Sipwise Ssip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and highly performant system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The Sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE can provide secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can, therefore, act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signalling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
What is new in Sip:provider v2.6?
There are a whole lot of new features, improvements and fixes since v2.5, here is an overview of the most important ones:
- IM/Presence, Buddy List and Remote Client Provisioning Support:
- SIP/SIMPLE for server-side presence handling, supporting PUBLISH, SUBSCRIBE and NOTIFY.
- XCAP for server-side buddy list management (add, delete and authorise contacts in buddy list).
- Instant Messaging (page mode supporting MESSAGE).
- Full support for the Jitsi multi-platform SIP client, including encrypted voice, video, instant messages, screen sharing, remote desktop control and contact lists.
- Remote Client Provisioning for Jitsi.
- VoIP Analyzer and Statistics (sip:providerPRO only):
- Passive capturing of any UDP and TCP-based SIP messages.
- Graphing of SIP signalling call flows in the Admin Panel.
- SIP Packet Inspection in the Admin Panel.
- SIP Packet Statistics.
- Media Features:
- Domain- and subscriber-specific sound sets for various announcements like caller busy, caller/callee locked, caller/callee blocked, number unavailable etc.
- Domain- and subscriber-specific Music on Hold (MoH) (sip:providerPRO only)
- System Enhancements:
- Full High-Availability of Media Streams during fail-over to the standby node (sip:providerPRO only)
- Full rolling-release support with config- and DB-schema management making platform upgrades starting from 2.6 to any future version a breeze.
- Fax Server Configuration in the Customer Self Care Panel (sip:providerPRO only)
- TCP and TLS Support for Peering Servers.
- Control Session-Timer settings per peer, domain and subscriber.
- Control user-provided and network-provided number handling for inbound calls per peer, domain and subscriber via the Admin panel.
- Control the content of various CLI related fields (From-Display, From-User, P-Preferred-Identity, P-Asserted-Identity) per peer, domain and subscriber via the Admin panel.
- Improved Emergency Call handling with full control over number normalisation via Rewrite Rules.
- Security Handling to view and manage blocked Subscribers and IPs via the Admin Panel.
- The configuration of Trusted Subscribers to allow unauthenticated calls via the Admin Panel.
- Disable number normalisation in Admin Panel, everything is expected to be entered as E.164 for better transparency.
- Upgrade of Kamailio to version 3.3.
- Multi-Threading support in media proxy-ng
- Fixed Voicebox folder handling.
- Fixed false-positive DDoS bans.
- Fixed empty NCOS list handling.
- Fixed WSDL errors for certain browsers.
- Fixed triggers in Peer handling.
- and dozens more…
How do I test-drive the new version?
If you want to test the Sip:provider together with the Jitsi client for getting all the shiny unified communication features, make sure to use the latest nightly build of Jitsi, otherwise it will most likely not work as expected!
The Sip:provider also supports Jitsi auto provisioning, the URL is:
How do I install the new version or upgrade from an older one?
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE v2.6 from scratch.
For users of the SPCE v2.5, please follow the upgrade procedure outlined in the Handbook. If you have customised your configurations using custom tt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise, all your calls will most certainly fail.
I want to thank our PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. I hope you enjoy using the v2.6 release and keep your input coming. A big thank you also to all the developers of Kamailio and SEMS, who make it possible for us to provide an innovative and future-proof SIP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise developers Mika, Andrew, Jon, Richard, Christian and Min, who worked insanely hard to create this release. You are plain awesome!