Thank you for the reply, it makes sense. We're using a line like this for
calls from plain RTP to SRTP, however the SDP arrives at the TLS phone with
no mention of encryption. Have you any idea what's wrong?
rtpengine_manage( "force trust-address replace-origin
replace-session-connection rtcp-mux-accept rtcp-mux-offer ICE=force
I've also attached the rtpengine log in case it helps.
Thanks very much.
On 27 July 2017 at 23:30, Richard Fuchs <rfuchs(a)sipwise.com> wrote:
On 07/27/2017 12:01 AM, David Cunningham wrote:
Thanks very much for that reply. We now detect whether the destination is
using TLS successfully using $ru and pcre_match().
Now when we call Asterisk -> Kamailio+rtpengine -> TLS phone, the TLS
phone rings but the call drops immediately when it answers. The issue is
that Asterisk doesn't like the 200 OK from the phone, which contains SRTP
information. The error logged by Asterisk is "Rejecting secure audio stream
without encryption details". I've included the SDP below.
Our questions now are:
1) Our goal is to have Kamailio+rtpengine act as a TLS/SRTP <--> Plain
SIP/RTP bridge. Is it possible to configure Kamailio so that Asterisk never
sees the encryption information in the 200 OK?
Yes, you just need to instruct rtpengine to translate the SDP to plain RTP
when sending to Asterisk. The appropriate flag to use in this case would be
`RTP/AVP`. Other flags might be relevant (e.g. if Asterisk doesn't want to
see any ICE information, use `ICE=remove`).
2) Is there anything wrong with the SDP returned by the TLS phone? For
example, you mentioned before SDES SRTP and I
wonder if the type of SRTP is
not acceptable for some reason.
This is also something you can control with flags given to rtpengine in
the other direction (plain RTP being translated to SRTP). By default, both
SDES and DTLS are offered. Either can be disabled by `SDES-off` and
`DTLS=off` respectively. Please see the docs at https://goo.gl/ivMQ6C
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