Differences between revisions 7 and 9 (spanning 2 versions)
Revision 7 as of 2014-01-17 17:22:50
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Revision 9 as of 2014-01-17 21:56:33
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   /!\ Don't do that. We don't want to couple service names with hosts names! -- zobel
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Just take it from [[https://github.com/opentelecoms-org/jscommunicator|upstream github]], use [[http://packages.debian.org/jscommunicator-web-phone|the web-phone package]] and symlink the files to your own web directory or clone an existing site: A live demo customised for the Debian community is at [[https://rtc.debian.org|rtc.debian.org]]

For those who want to host their own version and customise it to let their friends call them directly, j
ust take it from [[https://github.com/opentelecoms-org/jscommunicator|upstream github]], use [[http://packages.debian.org/jscommunicator-web-phone|the web-phone package]] and symlink the files to your own web directory or clone an existing site:

Key details

  • Debian Developers have access to the following services

    • SIP Proxy
    • TURN server
  • 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

Note:

  • 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

Lumicall

Lumicall is an open source mobile SIP client for Android. It only supports audio and does not support video or messaging yet. It has good support for SDES and ZRTP encryption, uses SIP over TLS and supports ICE and TURN for NAT busting.

  • Settings
    • SIP Identities (Add...)
      • SIP address/URI: username@debian.org

      • Profile enabled: Tick
      • Security mode: ZRTP
      • Gateway to PSTN: Remove tick
      • Intl. dialing prefix: 00
      • Authorization username: username@debian.org

      • Password: (your RTC password)
      • Registration: Tick
      • Use outbound proxy: Tick
      • Use STUN/TURN protocols: Tick
      • STUN server name: vogler.debian.org
      • STUN server port: 3478
      • STUN server protocol: udp

JSCommunicator

A live demo customised for the Debian community is at rtc.debian.org

For those who want to host their own version and customise it to let their friends call them directly, just take it from upstream github, use the web-phone package and symlink the files to your own web directory or clone an existing site:

mkdir /var/www/jscommunicator
cd /var/www/jscommunicator
wget -nH -r https://rtc.debian.org
vi config.js