Richard Fuchs writes:
You only need to specify the transport protocol in an answer if you want to change it from whatever the offering client sent (for whatever reason). If you don't specify the protocol in the answer (and you essentially did that as your flag wasn't understood and ignored), it will reply with the same protocol that the offering client used, which is normally what you want.
nice to hear. it simplifies my kamailio config a bit.
FTR, RFC 5764 states that "UDP/TLS/..." must be used when DTLS-SRTP is used, only WebRTC doesn't seem to honour that and omits this prefix, possibly because SDES exists (or used to exist) as an alternative to DTLS-SRTP within WebRTC. Which actually makes me wonder if WebRTC clients actually understand the UDP/TLS/... protocols...
based on my tests, jssip at least does not understand UDP/TLS/..., which is causing trouble. in order to make the mess bigger, looks like sdp will be dropped from next generation webrtc altogether (see http://ortc.org). this makes me wonder if it makes sense at all to try to support webrtc clients in a sip proxy.
-- juha