Thank you very much Arsen.
Knowing that this was supposed to work was what I needed to know. I was
only missing the NAT rule back to my phone system on the TLS port, which
is where the BYE packet was being sent to.
All works perfectly now.
On 9/19/20 4:05 AM, Arsen Semenov wrote:
Hello Conrad,
Bit hard to say exactly without looking into logs/dumps but it seems
like your call is long enough so router (if you have one) could
timeout on nat tcp connection, thus the “bye” request can not reach
uac. Just guessing.
Please check whether you have tcp keepalive enabled, example:
tcp_keepalive=yes
tcp_crlf_ping=yes
tcp_keepcnt=3
tcp_keepidle=30
tcp_keepintvl=30
Hope it helps.
On Sat, 19 Sep 2020 at 8:59 AM, conradcorde...@gmail.com
<mailto:conradcorde...@gmail.com> <conradcorde...@gmail.com
<mailto:conradcorde...@gmail.com>> wrote:
Hello,
Thank you for your reading this and for your help.
I'm a Kamailio newbie and managed to set up an edge proxy, which
works
perfectly on UDP traffic. I'm now attempting to deploy TLS/SRTP and
everything almost works perfectly. The single issue I'm having is
that
when either of the parties to an SRTP/TLS call disconnect, the other
party's call remains active. With UDP, when one of the parties
disconnects the call, the other leg of the call receives the BYE
command
and the call automatically disconnects.
This is how I have our infrastructure set up:
1. Twilio SIP Trunk with Secure Media enabled.
2. Kamailio 5.4.1 set up with TLS module, set to listen on TLS port
5061, SSL certificates from Let's Encrypt, route set to our phone
system.
3. Phone system is Asterisk.
As per above, everything works almost perfectly with TLS/SRTP. The
only
issue is that calls will not disconnect when one of the sides hang
up.
If I disable TLS/SRTP and use UDP only, everything works.
Audio is just fine with TLS/SRTP.
Does anyone know why calls with SRTP/TLS will not disconnect
automatically when one of the parties ends the call?
Thank you,
Conrad
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