[SR-Users] SRTP/TLS BYE Problem with One of the Legs of the Call

Arsen Semenov arsperger at gmail.com
Sat Sep 19 10:05:13 CEST 2020


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, conradcordeiro at gmail.com <
conradcordeiro at 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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>
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>
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>
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