A number of potential Summer of Code students are now looking at the question of how to improve RTC in Debian. Making RTC available as a service to the Debian community may be part of the solution, as this would give the community a foundation to test things against and a useful tool for communication.
One core component would be a TURN server to help people traverse NAT. There are two free TURN servers already packaged in Debian, and a third one that will potentially be packaged in the near future. Challenge for SoC students: how to make the TURN authentication mechanism work with the existing passwords of people with Debian accounts or alioth accounts? All of the TURN servers would require some code changes or development of new administration tools to link the authentication systems and support LDAP.
Another component would be a SIP or XMPP proxy. A very basic implementation would need to allow anybody with a debian.org account to register, make and receive calls, send and receive chat messages. Most SIP proxies and XMPP servers can do this. Challenges for SoC students: how to link up the authentication systems for user passwords? How/where can Debian/alioth users set preferences for call forwarding (adapting the code for the web interface)? How can these tools be integrated with other infrastructure such as the bug tracking system or IRC?
There are many additional services that are possible with RTC. A common example is conference calling. Challenge for students: how to set up one of the free conferencing solutions so that Debian Developers can quickly set up conferences using SIP, XMPP or WebRTC? Would it be administered by a web interface? Would it generate emails inviting people to join a conference with a WebRTC link to click?
Having RTC services always available on debian.org would enable RTC/VoIP packages to be automatically tested against the central infrastructure using the Jenkins continuous integration system. This would ensure that packages are always interoperable. Challenges for students: pick some of the packages and develop unit testing code for them that allows end-to-end tests against any other VoIP client software. Make it all run under Jenkins.