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| * SIP ([[http://tools.ietf.org/html/rfc3261|Session Initiation Protocol]]) Proxy, allowing to pass VoIP phone calls. decommissioned (https://alioth-lists.debian.net/pipermail/debian-rtc-team/2020-May/000194.html) * TURN ([[http://tools.ietf.org/html/rfc5766|Traversal Using Relays around NAT]]) server. decommissioned (https://alioth-lists.debian.net/pipermail/debian-rtc-team/2020-May/000194.html) |
* SIP ([[http://tools.ietf.org/html/rfc3261|Session Initiation Protocol]]) Proxy, allowing to pass VoIP phone calls. ([[https://alioth-lists.debian.net/pipermail/debian-rtc-team/2020-May/000194.html|decommissioned]]) * TURN ([[http://tools.ietf.org/html/rfc5766|Traversal Using Relays around NAT]]) server. ([[https://alioth-lists.debian.net/pipermail/debian-rtc-team/2020-May/000194.html|decommissioned]]) |
Key details
Please also see the Frequently Asked Questions
Debian Developers have access to the following services
SIP (Session Initiation Protocol) Proxy, allowing to pass VoIP phone calls. (decommissioned)
TURN (Traversal Using Relays around NAT) server. (decommissioned)
- XMPP/Jabber server
- The following services are NOT available yet but are being planned or under consideration
- BOSH Support, to connect to the XMPP server over HTTPS
You must create a real-time communications (RTC) password in the LDAP system: login to update your settings, then fill the RTC password field.
- Do not use the same password that you use for any other Debian service. For example, you may want to cache the RTC password in a mobile device where there is a risk that it will be compromised, exposing the password.
- Wait 30 minutes for the password to become active.
Your debian.org email address is also your XMPP address
E.g. username@debian.org
Contact and support
- For general questions about individual softphones, please contact the maintainers or upstream mailing lists
For general discussion about how to best use SIP as a tool to achieve the wider objectives of the Debian Project, please use debian-devel
For specific faults with the service, please contact the Debian Real Time Communications team
Instructions for various client programs
Conversations.im on Android
See this step by step guide.
Linphone configuration
Linphone seems to work well and can receive calls from Lumicall users
See the Linphone screenshots for full details
Jitsi configuration
Jitsi is one of the most extensive open source communications tools
See the Jitsi screenshots for full details
Empathy
- Empathy is the default communications client in the Gnome desktop
See the Empathy screenshots for full details
Lumicall
Lumicall is an open source mobile SIP client for Android.
See the Lumicall configuration page for full details
CSipSimple
CSipSimple integrates well with Android phones, and is recommended by The Guardian Project.
See the CSipSimple configuration page for full details
JSCommunicator
JSCommunicator is a browser-based WebRTC softphone using HTML5/JavaScript. It requires a modern browser.
A live demo customised for the Debian community is at rtc.debian.org
See the JSCommunicator configuration page for full details about how to put it in your own blog or web site
Asterisk
http://packages.debian.org/asterisk is a full PBX system.
See the Asterisk configuration page for full details
Ekiga
According to Wikipedia, ekiga does not currently support TLS connections.