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Sipwise C5 mr11.0.1 Released

We are excited to announce the general availability of the Sipwise C5CE and C5PRO mr11.0.1 release.

What is the Sipwise C5 platform?

Sipwise C5 PRO Architecture Overview

The Sipwise C5 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 high-performance 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 Sipwise C5 platform is available as a Community Edition (CE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup. The Sipwise C5 provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, 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 signaling 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’s new in mr11.0.1?

The most important changes for mr11.0 compared to mr10.5 are:

  • Discontinue iNew dedicated features and code [TT#179901]:

    • removed ‘config.yml’ section b2b.prepaid.inew

    • removed libinewrate selection for prepaid rating

    • discontinued libinewrate package.

    • removed ‘config.yml’ section apps.party_call_control

    • removed party_call_control subscriber’s preference

  • New option added in ‘config.yml’: kamailio.proxy.pbx.enable_cloud_pbx_hunt_timeout_for_parallel
    By default, this option is set to “yes” therefore, from the current version, the ringing duration for parallel group is defined by the setting PBX Group → Hunting Timeout (formerly named Serial Hunting Timeout) as for the other types of ringing policies.
    In previous versions, the duration of calls directed to parallel Hunt Groups were defined, as for normal calls to subscribers, by kamailio ‘fr_inv_timer’ (configurable by ‘config.yml’ option).
    In case you prefer the previous behavior, please set enable_cloud_pbx_hunt_timeout_for_parallel to “no”. [TT#180900]

  • Support measuring and reporting additional RTP/VoIP metrics, in particular for scenarios that involve only plain RTP passthrough. [TT#178352]

  • Added support for g722 and g729 codecs on Asterisk. This allows Asterisk to use them on voicemail [TT#187100].

  • New option added to ‘config.yml’: asterisk.sip.codecs. It indicates the list of codecs, in order of importance, to be used on Asterisk. The available options, separated by commas, are ‘g729’, ‘g722’, ‘opus’, ‘alaw’ and ‘ulaw’. [TT#187100]

  • Add setting to control the Opus encoder complexity for RTP transcoding scenarios (opus_complexity in preferences for subscribers, peers, and domains). The default setting is to achieve the highest encoder quality. With the new setting a lower quality can be selected, saving CPU time as a trade-off. [TT#185100]

  • New option added to ‘config.yml’: It contains the list of SIP methods that are skipped by kamailio topos module. By default it includes only ‘OPTIONS’ method. [TT#180650]

  • Add support for a new type of hosts dedicated to rtp processing. The new host name is rtp followed as by convention by 2 digits and a letter (e.g. rtp01a). [TT#185050]

  • Add suffix to emergency mapping. [TT#96652]

  • Add support for rtpengine instances [TT#186700]

See the list of all changes in PDF Changelog_mr11.0.1

What is mr11.0.1?

The build mr11.0.1 is the first build for release mr11.0, mr11.0.1 provides new features and bugfixes.

Is mr11.0 LTS (long time supported) release?

No. See the release calendar on

How do I test-drive the new version?

As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. For those of you using Amazon Cloud we provide the EC2 AMIs in the following regions:

  • AMI ID for region eu-west-1: ami-08210c7f1164551d6

Check the relevant section in the Sipwise C5 CE Handbook for detailed instructions.

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 Sipwise C5 CE mr11.0.1 from scratch. For the users of the previous version of Sipwise C5 CE, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise, all your calls will most certainly fail. Hint: modern patchtt.tt2 framework can help you here in the future.

How can I contribute to the project?

Sipwise is publishing software components at Please check it regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our Dev Mailing-List.


We want to thank our PRO/Carrier customers and the Sipwise C5 CE community for their feedback,bug reports and feature suggestions to make this release happen. We hope you enjoy using our software and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!