Hi all,<br><br>I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at:<br><br><a href="https://supportforums.cisco.com/docs/DOC-15381">https://supportforums.cisco.com/docs/DOC-15381</a><br>
<br>That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically, this is what it said:<br><br>==============================================<br><i style="font-size:10pt;font-family:arial,helvetica,sans-serif"><b>Note</b>:
There is no optional SRTP mode in Asterisk, i.e. if encryption is
active on peer, it will not accept non-ciphered audio and viceversa. On
the IP phones, however, it is possible to have unsecure calls if the
other peer does not support SRTP, i.e. incoming calls may work, but not
outgoing calls. This is an Asterisk limitation (Snom supports also the
“optional”mode on SRTP sending two m=audio attributes, but Asterisk does
not know how to handle those descriptors).</i><br>==============================================<br><br>This is from a quite dated article (2011), so I'm hoping that I newer versions of Asterisk will fall back on plaintext if TLS isn't available for some reason.<br>
<br>Secondly, is there any way to detect if a call is secure from inside the dialplan or AGI script?<br><br>I think that's all for now.<br><br>Thanks in advance,<br><br>Mike Diehl.<br>