A few problems were observed during the demo of RTC services at the Cambridge mini-DebConf yesterday. As it turns out, many of them are already documented and solutions are available for some of them.
I had made some test calls on Friday using rtc.debian.org and I still had the site open in another tab in another browser window. When people tried to call me during the demo, both tabs were actually ringing but only one was visible.
When a SIP client registers, the SIP registration server sends it a list of all other concurrent registrations in the response message. We simply need to extend JSCommunicator to inspect the response message and give some visual feedback about other concurrent registrations. Issue #69. SIP also provides a mechanism to clear concurrent registrations and that could be made available with a button or configuration option too (Issue #9).
The second issue during the WebRTC demo was that the callee (myself) was alerted about the call before the ICE checks had been performed. The optimal procedure to provide a slick user experience is to run the connectivity checks before alerting the callee. If the connectivity checks fail, the callee should never be alerted with a ringing sound and would never know somebody had tried to call. The caller would be told that the call was unable to be attempted and encouraged to consider trying again on another wifi network.
RFC 5245 recommends that connectivity checks should be done first but it is not mandatory. One reason this is problematic with WebRTC is the need to display the pop-up asking the user for permission to share their microphone and webcam: the popup must appear before connectivity checks can commence. This has been discussed in the JsSIP issue tracker.
Non-WebRTC softphones, such as Lumicall, do the connectivity checks before alerting the callee.
It appears the corporate wifi network in the venue was blocking the UDP packets so the connectivity checks could never complete, not even using a TURN server to relay the packets.
People trying to use the service on home wifi networks, in small offices and mobile tethering should not have this problem as these services generally permit UDP by default.
Some corporate networks, student accommodation and wifi networks in some larger hotels have blocked UDP and in these cases, additional effort must be made to get through the firewall.
The TURN server we are running for rtc.debian.org also supports a TLS transport but it simply isn't configured yet. At the time we originally launched the WebRTC service in 2013, the browsers didn't support TURN over TLS at all but now they do. This is probably the biggest problem encountered during the demo but it does not require any code change to resolve this, just configuration, so a solution is well within reach.
During the demo, we worked around the issue by turning off the wifi on my laptop and using tethering with a 4G mobile network. All the calls made successfully during the demo used the tethering solution.
While these lengthy connectivity checks appear disappointing, it is worth remembering that this is an improvement over the first generation of softphones: none of them made these checks at all, they would simply tell the user the call had been answered but audio and video would only be working in one direction or not at all.
I'd like to thank all the participants in the demo, including Juliana Louback who called us from New York, Laura Arjona who called us from Madrid, Daniel Silverstone who called from about three meters away in the front row and Iain Learmonth who helped co-ordinate the test calls over IRC.
Thanks are also due to Steve McIntyre, the local Debian community, ARM and the other sponsors for making another mini-DebConf in the UK this year.