Hi, I realize Digium is not necessarily going to fix the DTLS-SRTP issues immediately
However, I think it would be really important to clarify the situation with the media sub-field for the protocol, if the current Asterisk behavior is not correct, then it would be good to tell people not to try and duplicate this behavior in other products that are intended to work with Asterisk Specifically, Asterisk is expecting "UDP/TLS/RTP/SAVPF" in the protocol sub-field of an INVITE message The RFC 5763 examples seem to show that the INVITE should just say "RTP/SAVPF" and the full value of "UDP/TLS/RTP/SAVPF" is only used in the SDP sent with a 200 OK response. Is there some other permutation of this RFC that Asterisk is aiming to adhere to? Or is the current behavior just a bug that other users should ignore and not try to emulate? Regards Daniel -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
