Tls / srtp?

Hi,

I have setup Asterisk 10.5 and eyeBeam 1.5.9 for TLS / SRTP. When making calls, “Lock” icon (in locked mode) appears in eyeBeam showing call is using TLS / SRTP.

However in Asterisk CLI, it shows

– Registered SIP ‘1001’ at xxx.xxx.89.9:3065
== Using SIP RTP CoS mark 5
– Executing [201@myphones:1] Answer(“SIP/1001-00000021”, “”) in new stack

I would like to ask,

  1. Why is eyeBeam registering at port 3065 while the default port for TLS connection should be 5061 ?
  2. Why does CLI saying “Using SIP RTP CoS…” instead of “Using SIP SRTP CoS…”
  3. How to check inside asterisk cli if an ongoing call is TLS / SRTP or not ?

Thanks,
Kind

Anyone ?!

  1. This will be the source port number.

  2. QoS is implemented at the network layer. The code of this is probably in common with that for simple RTP. Remember this is only an informative message.

  3. The most reliable way of confirming is to run something like wireshark and checking what is going over the wire. Not using TLS/SRTP myself, I haven’t looked deeply into the diagnostics provided by Asterisk.

Have you tried turning up the chan_sip debugging level?

If all else fails, you can modify the source code to report the information you want. This should be a simple and safe change.

If the softphone is on the same system, it may not be able to bind 5061.