jan.gaida at grupoampe...
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 Posted: Tue Jun 10, 2014 10:28 am Post subject: [asterisk-users] SIP call control via RTCP |
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Hello,
I have found here 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@grupoamper.com (jan.gaida@grupoamper.com)> wrote:
| Quote: | 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@zigg.com (matt@zigg.com)> wrote:
| Quote: | On May 12, 2014, at 5:02 AM, Jan Gaida <jan.gaida@grupoamper.com (jan.gaida@grupoamper.com)> wrote:
| Quote: | 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?
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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@grupoamper.com (jan.gaida@grupoamper.com) | www.grupoamper.com 
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Jan Gaida
Ingeniero Desarrollo Software C/ Marconi 3 (PTM)
28760 Tres Cantos
Spain
jan.gaida@grupoamper.com (jan.gaida@grupoamper.com) | www.grupoamper.com
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