27 Jul
2017
27 Jul
'17
2:30 p.m.
On 07/27/2017 12:01 AM, David Cunningham wrote:
Hi Daniel,
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
Cheers