[asterisk-dev] DTLS-SRTP SDP correction?

[Daniel Pocock]

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