[asterisk-users] SIP call control via RTCP

[Jan Gaida]

Hello,

I have found here <http://www.voip-info.org/wiki/view/Asterisk+RTCP> that
there has been a patch for RTCP of Asterisk 1.4.

Does this mean that starting with Asterisk version 1.6 RTCP call control is
working correctly?

Kind regards
Jan Gaida


On Mon, May 12, 2014 at 2:40 PM, Jan Gaida <jan.gaida at grupoamper.com> wrote:

> 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
>



-- 
    *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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