[asterisk-users] SIP call control via RTCP

Jan Gaida jan.gaida at grupoamper.com
Mon May 12 07:40:45 CDT 2014


Thank you.
Yes, that should work. But if I understand it correctly, only if there's no
silence detection activated. Otherwise, when silence is detected no RTP
would be send, so that rtptimeout would hang up a still active call.

I there no option to use RTCP? Not even in Asterisk 11?

Regards


On Mon, May 12, 2014 at 2:12 PM, Matt Behrens <matt at zigg.com> wrote:

> On May 12, 2014, at 5:02 AM, Jan Gaida <jan.gaida at grupoamper.com> wrote:
>
> > We are using Asterisk 1.4 as call distribution system with simple queues
> for SIP calls.
> >
> > With high load (4000 calls/hour) some calls remain in queue forever
> (until queue's max wait time) in spite of being hung up already by the
> caller.  It seems that when a BYE is lost, Asterisk has no mechanism to
> check whether a call is still active.
> >
> > Is there a way to activate a RTCP call control, e.g. Asterisk should
> hang up when he stops receiving RTCP messages?
>
>
> Have you looked at the rtptimeout and rtpholdtimeout options?
>
>
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-- 
    *Jan **Gaida*
Ingeniero Desarrollo Software     C/ Marconi 3 (PTM)
28760 Tres Cantos
Spain
jan.gaida at grupoamper.com | www.grupoamper.com

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