Hello,
for phones that are using tls, you can do the following tests:
- for incoming traffic: proto==TLS
- for outgoing traffic: after lookup location, the R-URI ($ru) should
have 'transport=tls'
For RTPEngine there are some flags to specify you want or not SDES SRTP,
I used them few times in the past, but I don't recall them by heart --
the docs should have them.
Cheers,
Daniel
On 26.07.17 06:40, David Cunningham wrote:
Hello,
We're configuring Kamailio 4.2 with rtpengine to act as a midpoint
between a telephone using TLS/SRTP and Asterisk. There are examples
out there for TLS/SRTP with WebRTC, but we're using a plain hard
phone, not WebRTC.
Would anyone be able to point us towards a Kamaiio configuration which:
a) Tests if the destination phone (stored using usrloc) uses TLS.
b) Sends RTP for calls to a TLS phone to rtpengine to be encrypted. We
can assume all phones using TLS want to use SRTP.
Thanks very much in advance.
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