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 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
