Differences between revisions 27 and 31 (spanning 4 versions)
Revision 27 as of 2016-04-10 12:18:26
Size: 7050
Editor: ?AndreasMundt
Comment: update
Revision 31 as of 2020-05-08 12:05:46
Size: 3993
Deletions are marked like this. Additions are marked like this.
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 * To start quickly, just skip below to the client configuration links
  * Using the browser-based WebRTC service is the quickest way to start
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  * SIP ([[http://tools.ietf.org/html/rfc3261|Session Initiation Protocol]]) Proxy, allowing to pass VoIP phone calls,
  * TURN ([[http://tools.ietf.org/html/rfc5766|Traversal Using Relays around NAT]]) server
  * 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)
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 * The following services are NOT available yet but are being planned or under consideration
  * Audio and/or video conference software
  * [[http://packages.debian.org/reconserver|see the reConServer package]]
   * [[https://jitsi.org/videobridge|Jitsi video bridge]] is also cool
 * The following services are NOT available yet but are being planned or under consideration 
  * BOSH Support, to connect to the XMPP server over HTTPS
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 * 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
 * If you are lucky, your client software uses DNS NAPTR and SRV lookups to find the TURN and SIP servers, if not, you can hardcode the following values into your configuration:
  * SIP server: please use a client that finds it using SRV lookups
   * if really necessary, use the value ''vogler.debian.org'' and TLS, port 5061
   /!\ Don't do that. We don't want to couple service names with hosts names! -- zobel
  * TURN server: ''turn.debian.org'' (UDP port 3478)
  * SIP over WebSocket: ''sip-ws.debian.org'' (HTTP port 443)
 * [[http://www.sip5060.net|sip5060.net]] provides some [[http://www.sip5060.net/test-calls|convenient SIP test numbers]]
  * clicking the links on the [[http://www.sip5060.net/test-calls|sip5060.net site]] will dial through your local SIP client or [[https://freephonebox.net|FreePhoneBox.net]]
  * copy and paste the SIP addresses from the page into [[https://rtc.debian.org|rtc.debian.org]] to test from there
 * Your ''debian.org'' email address is also your XMPP address
  * E.g. ''username@debian.org''
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 * For specific faults with the service, please contact the [[Teams/DSA|Debian System Administration (DSA)]] team  * For specific faults with the service, please contact the [[Teams/RTC|Debian Real Time Communications]] team
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== NAT traversal ==

 * 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 [[https://rtc.debian.org|rtc.debian.org]] WebRTC service is based on [[http://jscommunicator.org|JSCommunicator]]. It supports 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.

== WebRTC status ==


 * [[http://www.webrtc.org/|WebRTC]] users can only interact with other WebRTC users
 * Jitsi and Lumicall users can interact with each other but not with WebRTC users (yet)
 * People who don't have a SIP account can call you instantly using the [[https://freephonebox.net|FreePhoneBox.net WebRTC service]]
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=== Conversations.im on Android ===

See this [[https://wiki.debian.org/UnifiedCommunications/DebianDevelopers/UserGuide/Android|step by step guide]].

Key details

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

Jitsi configuration

  • Jitsi is one of the most extensive open source communications tools

  • See the Jitsi screenshots for full details


  • Empathy is the default communications client in the Gnome desktop
  • See the Empathy screenshots for full details






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