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Comment: added instructions for client app conversations.im on android
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  * E.g. ''pocock@debian.org''   * E.g. ''username@debian.org''
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  * Put your full SIP address, e.g. ''pocock@debian.org'' in this field   * Put your full SIP address, e.g. ''username@debian.org'' in this field
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   * 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
   * if really necessary, use the value ''sip-ws.debian.org'' and TLS, port 5061
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=== Conversations.im on Android ===

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

Key details

  • To start quickly, just skip below to the client configuration links
    • Using the browser-based WebRTC service is the quickest way to start
  • 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
  • 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 SIP address

  • 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. username@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 sip-ws.debian.org and TLS, port 5061

    • TURN server: turn.debian.org (UDP port 3478)

    • SIP over ?WebSocket: sip-ws.debian.org (HTTP port 443)

  • sip5060.net provides some convenient SIP test numbers

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 System Administration (DSA) team

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 rtc.debian.org WebRTC service is based on 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


  • 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 FreePhoneBox.net WebRTC service

Instructions for various client programs

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.

Conversations.im on Android

See this step by step guide.