|
Size: 4001
Comment: use links for decommision mails
|
Size: 3875
Comment: cleanup
|
| Deletions are marked like this. | Additions are marked like this. |
| Line 2: | Line 2: |
| Line 6: | Line 5: |
| * 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]]) |
* --(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]]) |
| Line 11: | Line 10: |
| * You must create a real-time communications (RTC) password in [[http://db.debian.org|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. |
* You must create a real-time communications (RTC) password in [[https://db.debian.org|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 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. |
| Line 20: | Line 19: |
| * 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 general discussion about how to best use these technologies as a tool to achieve the wider objectives of the Debian Project, please use ''debian-devel'' |
| Line 22: | Line 21: |
| Line 28: | Line 26: |
| See this [[https://wiki.debian.org/UnifiedCommunications/DebianDevelopers/UserGuide/Android|step by step guide]]. | See this [[UnifiedCommunications/DebianDevelopers/UserGuide/Android|step by step guide]]. |
| Line 32: | Line 30: |
| * [[http://packages.debian.org/linphone|Linphone]] seems to work well and can receive calls from [[http://www.lumicall.org|Lumicall]] users | * [[DebianPackage:linphone|Linphone]] seems to work well and can receive calls from [[https://lumicall.org/|Lumicall]] users |
| Line 34: | Line 32: |
| Line 38: | Line 35: |
| * [[http://packages.debian.org/jitsi|Jitsi]] is one of the most extensive open source communications tools | * [[DebianPackage:jitsi|Jitsi]] is one of the most extensive open source communications tools, but is no longer available in Debian |
| Line 43: | Line 40: |
| * Empathy is the default communications client in the Gnome desktop | * Empathy is the default communications client in the GNOME desktop |
| Line 48: | Line 45: |
| * [[http://www.lumicall.org|Lumicall]] is an open source mobile SIP client for Android. | * [[https://lumicall.org/|Lumicall]] is an open source mobile SIP client for Android. |
| Line 52: | Line 49: |
| * [[https://code.google.com/p/csipsimple/|CSipSimple]] integrates well with Android phones, and is recommended by [[https://guardianproject.info/apps/|The Guardian Project]]. | * [[WikiPedia:CSipSimple|CSipSimple]] was a project that integrated well with Android phones, and was recommended by [[https://guardianproject.info/apps/|The Guardian Project]]. |
| Line 57: | Line 55: |
| * [[http://jscommunicator.org|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 [[https://rtc.debian.org|rtc.debian.org]] |
* [[https://jscommunicator.org/|JSCommunicator]] is a browser-based WebRTC softphone using HTML5/JavaScript. It requires a modern browser. |
| Line 63: | Line 60: |
| * [[http://packages.debian.org/asterisk]] is a full PBX system. | * [[DebianPackage:asterisk]] is a full PBX system. |
| Line 68: | Line 65: |
| According to [[http://en.wikipedia.org/wiki/Comparison_of_VoIP_software|Wikipedia]], ekiga does not currently support TLS connections. | According to [[WikiPedia:Comparison_of_VoIP_software|Wikipedia]], ekiga does not currently support TLS connections. |
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 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 these technologies 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, but is no longer available in Debian
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 was a project that integrated well with Android phones, and was 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.
See the ?JSCommunicator configuration page for full details about how to put it in your own blog or web site
Asterisk
Ekiga
According to Wikipedia, ekiga does not currently support TLS connections.