Differences between revisions 3 and 33 (spanning 30 versions)
Revision 3 as of 2014-01-17 08:27:42
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Editor: DanielPocock
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Revision 33 as of 2021-06-25 01:50:42
Size: 3875
Editor: PaulWise
Comment: cleanup
Deletions are marked like this. Additions are marked like this.
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   * Please also see the [[UnifiedCommunications/DebianDevelopers/FAQ|Frequently Asked Questions]]
 * [[DebianDeveloper|Debian Developers]] have access to the following services
  * --(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]])
  * 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 [[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.
  * 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 ==
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 * [[DebianDeveloper|Debian Developers]] have access to the following services
  * SIP Proxy
  * TURN server
 * You must create a real-time communications (RTC) password in [[http://db.debian.org|the LDAP system]]
  * 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.
  * Wait 30 minutes for the password to become active.
 * Your ''debian.org'' email address is also your SIP address
  * E.g. ''pocock@debian.org''
 * Your SIP software may try to use the user-part of the SIP address for authentication. It will not work.
  * In your SIP settings, look for an ''authentication username'' or ''auth user'' field. It is often blank by default.
  * Put your full SIP address, e.g. ''pocock@debian.org'' in this field
 * The same credentials are used for TURN
 * 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 [[Teams/RTC|Debian Real Time Communications]] team
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== NAT traversal == == Instructions for various client programs ==
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 * NAT and firewalls have traditionally been a problem for free RTC software
 * For SIP itself, we only use TLS
  * This is a stream connection that is more likely to get through NAT than UDP
  * It can also potentially be tunnelled through proxies using the HTTP CONNECT method (port 443)
  * Some routers try to mangle SIP packets to help them through NAT, in practice this sometimes makes the problem worse
  * By using TLS, we ensure that no intermediate device will tamper with the packets, we aim to use industry standard ICE and TURN
 * The modern approach to this problem is the use of Internet Connectivity Establishment (ICE) and, as a last resort, relaying traffic through a TURN server
 * Not all SIP clients support TURN
  * Jitsi only supports TURN with Jabber, the SIP-TURN support is coming
  * Empathy only supports TURN through Google's proprietary TURN servers, but the TURN code could use any TURN server if configuration options were available. There is a bug report for this.
  * Only one end of the connection needs a TURN server for it to work though, as long as both support ICE.
 * The WebRTC demo site based on [[http://jscommunicator.org|JSCommunicator]] does support both ICE and TURN and is pre-configured for Debian's TURN servers. Although the UI is very basic, there is a high probability that it can get through NAT in situations where the other SIP clients currently struggle.
=== Conversations.im on Android ===
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== Jitsi configuration == See this [[UnifiedCommunications/DebianDevelopers/UserGuide/Android|step by step guide]].
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=== Linphone configuration ===

 * [[DebianPackage:linphone|Linphone]] seems to work well and can receive calls from [[https://lumicall.org/|Lumicall]] users
 * See the [[UnifiedCommunications/DebianDevelopers/UserGuide/Linphone|Linphone screenshots]] for full details

=== Jitsi configuration ===

 * [[DebianPackage:jitsi|Jitsi]] is one of the most extensive open source communications tools, but is no longer available in Debian
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=== Empathy ===

 * Empathy is the default communications client in the GNOME desktop
 * See the [[UnifiedCommunications/DebianDevelopers/Empathy|Empathy screenshots]] for full details

=== Lumicall ===

 * [[https://lumicall.org/|Lumicall]] is an open source mobile SIP client for Android.
 * See the [[UnifiedCommunications/DebianDevelopers/UserGuide/Lumicall|Lumicall configuration page]] for full details

=== CSipSimple ===

 * [[WikiPedia:CSipSimple|CSipSimple]] was a project that integrated well with Android phones, and was recommended by [[https://guardianproject.info/apps/|The Guardian Project]].
 * See the [[/CSipSimple|CSipSimple configuration page]] for full details

=== JSCommunicator ===

 * [[https://jscommunicator.org/|JSCommunicator]] is a browser-based WebRTC softphone using HTML5/JavaScript. It requires a modern browser.
 * See the [[UnifiedCommunications/DebianDevelopers/UserGuide/JSCommunicator|JSCommunicator configuration page]] for full details about how to put it in your own blog or web site

=== Asterisk ===

 * [[DebianPackage:asterisk]] is a full PBX system.
 * See the [[UnifiedCommunications/DebianDevelopers/UserGuide/Asterisk|Asterisk configuration page]] for full details

=== Ekiga ===

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

  • 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

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

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

CSipSimple

JSCommunicator

Asterisk

Ekiga

According to Wikipedia, ekiga does not currently support TLS connections.