[asterisk-dev] PJSIP and persistent TLS
Joshua Colp
jcolp at digium.com
Mon Mar 3 11:54:14 CST 2014
On 14-03-03 12:51 PM, André Valentin wrote:
> Hi!
Hola,
> I'm just trying to move my function ality from chan_sip to pjsip. I
> stumbled upon one problem. With chan_sip and a via persistant TLs
> connected phone everything works as expected. Calls in/out work. Even
> if asterisk tries to reach the phone, it reuses the existing TLS
> connection.
>
> If I switch this to PJSIP, it stops working. I configured the
> following parameters: symmetric_rtp=true force_rport=true and
> others...
>
> I I know call the phone via PJSIP, asterisk does not reuse the TLS
> connection. It tries to create a new one, which of course fails.
>
> Any ideas?
What's the exact configuration in use? Do you have a transport
explicitly specified for the endpoint? Doing so will currently cause it
to try to create a new connection [1].
[1] https://issues.asterisk.org/jira/browse/ASTERISK-22658
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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