Differences between revisions 16 and 18 (spanning 2 versions)
Revision 16 as of 2014-01-19 13:35:44
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Comment: Add pointer for WebRTC
Revision 18 as of 2014-01-19 13:38:38
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  * SIP ([[http://tools.ietf.org/html/rfc3261|Session Initiation Protocol]]) Proxy
  * TURN server
  * 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

Key details

  • To start quickly, just skip below to the client configuration links
  • 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

    • 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. 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)

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)

Instructions for various client programs

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