I have a working telephone project that uses SIP.js 0.7.5 with Asterisk on the server side. Currently it handles both audio and video correctly.
The SIP.js webpage has instructions for setting up a datachannel through a SIP call. The online demo uses OpenSIPS.
When setting up a SIP call with a datachannel through the SIP.js online demo, the SDP looks like this:
o=- 2849011408178506361 2 IN IP4 127.0.0.1
m=application 45029 DTLS/SCTP 5000
c=IN IP4 220.127.116.11